February 2011 Archives by author

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Starting: Tue Feb 1 02:24:20 CST 2011
Ending: Mon Feb 28 23:20:03 CST 2011
Messages: 1077

• [asterisk-users] Problems with realtime sip   Fellipe ...
• [asterisk-users] DTMF not detected, time out   Fellipe ...
• [asterisk-users] SIP METHOD BYE   Fellipe ...
• [asterisk-users] Radius Based Accounting for Asterisk   Gopalakrishnan A.N
• [asterisk-users] PRI voice optimization   Gopalakrishnan A.N
• [asterisk-users] zaptel/dahdi settings for singtel E1 line   Gopalakrishnan A.N
• [asterisk-users] sangoma wanpipe install error   Gopalakrishnan A.N
• [asterisk-users] Trunk grouping   Gopalakrishnan A.N
• [asterisk-users] Newbie´s question about Asterisk...   Gopalakrishnan A.N
• [asterisk-users] RTP (voice) issue. STUN server   Gopalakrishnan A.N
• [asterisk-users] digium te220   Albert
• [asterisk-users] YELLOW alarm   Albert
• [asterisk-users] YELLOW alarm   Albert
• [asterisk-users] YELLOW alarm   Albert
• [asterisk-users] YELLOW alarm   Albert
• [asterisk-users] application for voice modulation   Albert
• [asterisk-users] calls are not going thru e1 line   Albert
• [asterisk-users] calls are not going thru e1 line   Albert
• [asterisk-users] application for voice modulation   Albert
• [asterisk-users] SIP Hardphone that works well with asterisk   Albert
• [asterisk-users] Dial() function   Albert
• [asterisk-users] Statistics for Asterisk CDR   Albert
• [asterisk-users] calls are not going thru e1 line   Albert
• [asterisk-users] AstriEurope coference   Albert
• [asterisk-users] AstriEurope coference   Albert
• [asterisk-users] calls are not going thru e1 line   Albert
• [asterisk-users] exceeds the maximum size of ast_fdset error on Asterisk-1.8.0   Benny Amorsen
• [asterisk-users] Hide the plain text password   Benny Amorsen
• [asterisk-users] DTMF and Snom   Benny Amorsen
• [asterisk-users] About maxlen parameter in queues   Daniel - Asterisk
• [asterisk-users] About maxlen parameter in queues   Daniel - Asterisk
• [asterisk-users] About maxlen parameter in queues   Daniel - Asterisk
• [asterisk-users] Assigning an extension to a roaming phone   Axelle
• [asterisk-users] Assigning an extension to a roaming phone   Axelle
• [asterisk-users] Assigning an extension to a roaming phone   Axelle
• [asterisk-users] Assigning an extension to a roaming phone   Axelle
• [asterisk-users] Assigning an extension to a roaming phone   Axelle
• [asterisk-users] Assigning an extension to a roaming phone   Axelle
• [asterisk-users] Assigning an extension to a roaming phone   Axelle
• [asterisk-users] Assigning an extension to a roaming phone   Axelle
• [asterisk-users] Registration failed though configured.   Axelle
• [asterisk-users] Assigning an extension to a roaming phone   Axelle
• [asterisk-users] Assigning an extension to a roaming phone   Axelle
• [asterisk-users] Registration failed though configured.   Axelle
• [asterisk-users] Any voice changer applications for Asterisk?   Bruce B
• [asterisk-users] Any voice changer applications for Asterisk?   Bruce B
• [asterisk-users] Can a duration limit be specified in spool call file?   Bruce B
• [asterisk-users] Can a duration limit be specified in spool call file?   Bruce B
• [asterisk-users] Any voice changer applications for Asterisk?   Bruce B
• [asterisk-users] Can a duration limit be specified in spool call file?   Bruce B
• [asterisk-users] Callback through extensions.conf?   Bruce B
• [asterisk-users] Callback through extensions.conf?   Bruce B
• [asterisk-users] Can a duration limit be specified in spool call file?   Bruce B
• [asterisk-users] IP ban list by country   Bruce B
• [asterisk-users] Asterisk Call File using Local Channel not passing Variable back to Dialplan   Bruce B
• [asterisk-users] Asterisk Call File using Local Channel not passing Variable back to Dialplan   Bruce B
• [asterisk-users] No ring tone on inbound call - but channel connects fine   Bruce B
• [asterisk-users] No ring tone on inbound call - but channelconnects fine   Bruce B
• [asterisk-users] PRI "wanrouter status" shows disconnected - system problem or Telco?   Bruce B
• [asterisk-users] YELLOW alarm   Sébastien BERGER
• [asterisk-users] YELLOW alarm   Sébastien BERGER
• [asterisk-users] YELLOW alarm   Sébastien BERGER
• [asterisk-users] T.38 negotiation error   David Backeberg
• [asterisk-users] Un message de Mickael t'attend...   Badoo
• [asterisk-users] Lis tes messages avant qu'ils ne soient effacés!   Badoo
• [asterisk-users] How to disable srtp in asterisk 1.8.2.3?   Miguel Baptista
• [asterisk-users] How to disable srtp in asterisk 1.8.2.3?   Miguel Baptista
• [asterisk-users] How to disable srtp in asterisk 1.8.2.3?   Miguel Baptista
• [asterisk-users] How to disable srtp in asterisk 1.8.2.3?   Bob Beers
• [asterisk-users] Question about EuroBRI final 2 digits   Bob Beers
• [asterisk-users] Playback in uplink and recording in downlink   Paul Belanger
• [asterisk-users] Return variables from func_odbc calls?   Paul Belanger
• [asterisk-users] Return variables from func_odbc calls?   Paul Belanger
• [asterisk-users] Callback through extensions.conf?   Paul Belanger
• [asterisk-users] Asterisk 1.8.3   Paul Belanger
• [asterisk-users] Asterisk 1.8.3   Paul Belanger
• [asterisk-users] Asterisk compile option DAHDI SPANS   Paul Belanger
• [asterisk-users] Asterisk compile option DAHDI SPANS   Paul Belanger
• [asterisk-users] Realtime and Local Channel Crash Problem 1.8.3-rc2   Paul Belanger
• [asterisk-users] Regarding Asterisk   Paul Belanger
• [asterisk-users] application for voice modulation   Paul Belanger
• [asterisk-users] Dial() function   Paul Belanger
• [asterisk-users] no progress indication   Paul Belanger
• [asterisk-users] AMI FullyBooted issue   Paul Belanger
• [asterisk-users] Unable to make outgoing calls with Internode   Dovid Bender
• [asterisk-users] Defining what an extension should do after theDial() command returns busy.   Dovid Bender
• [asterisk-users] dial option 'g' not working   Dovid Bender
• [asterisk-users] Assigning an extension to a roaming phone   Benoit
• [asterisk-users] downgrade libpri   Agustina Berretta
• [asterisk-users] [OT] Yealink IP Phones   Ivan Bolognani
• [asterisk-users] outbound call leg CALLID   Borin
• [asterisk-users] outbound call leg CALLID   Borin
• [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2   Borin
• [asterisk-users] lua -asterisk manual   Borin
• [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2   Borin
• [asterisk-users] lua -asterisk manual   Borin
• [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2   Borin
• [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2   Borin
• [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2   Borin
• [asterisk-users] lua -asterisk manual   Borin
• [asterisk-users] extensions.lua with luasql.mysql.   Borin
• [asterisk-users] RTP (voice) issue. STUN server   Oleg Botvinkin
• [asterisk-users] Pickup from an specific exten   Jordi Bou
• [asterisk-users] Paging a message. How?   Russell Brown
• [asterisk-users] SIP Hardphone that works well with asterisk   Terry Brummell
• [asterisk-users] SIP Hardphone that works well with asterisk   Terry Brummell
• [asterisk-users] Newbie´s question about Asterisk...   Terry Brummell
• [asterisk-users] Newbie´s question about Asterisk...   Terry Brummell
• [asterisk-users] Newbie´s question about Asterisk...   Terry Brummell
• [asterisk-users] Paging with Polycom 3.3.x   Terry Brummell
• [asterisk-users] Paging with Polycom 3.3.x   Terry Brummell
• [asterisk-users] Paging with Polycom 3.3.x   Terry Brummell
• [asterisk-users] asterisk security....again   Terry Brummell
• [asterisk-users] Erroneous email from JIRA   Russell Bryant
• [asterisk-users] unregistered trunks and registered phones coming from the same IP   Ricardo Carvalho
• [asterisk-users] trunks and phones registered from the same IP   Ricardo Carvalho
• [asterisk-users] trunks and phones registered from the same IP   Ricardo Carvalho
• [asterisk-users] unregistered trunks and registered phones coming from the same IP   Ricardo Carvalho
• [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP   Ricardo Carvalho
• [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP   Ricardo Carvalho
• [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP   Ricardo Carvalho
• [asterisk-users] asterisk security....again   Ricardo Carvalho
• [asterisk-users] SIP Originate on 1.8.X   Carlos Chavez
• [asterisk-users] Problems using Background within a macro on V 1.4   Carlos Chavez
• [asterisk-users] About maxlen parameter in queues   Carlos Chavez
• [asterisk-users] Defining what an extension should do after the Dial() command returns busy.   Carlos Chavez
• [asterisk-users] Error loading module ��Է�Vi.so   Carlos Chavez
• [asterisk-users] Pickup from an specific exten   Carlos Chavez
• [asterisk-users] Assigning an extension to a roaming phone   Carlos Chavez
• [asterisk-users] Asterisk Using as a SIP Client   Khaled W. Chehab
• [asterisk-users] Asterisk/Skype   Khaled W. Chehab
• [asterisk-users] Asterisk/Skype   Khaled W. Chehab
• [asterisk-users] Asterisk/Skype   Khaled W. Chehab
• [asterisk-users] Asterisk/Skype   Khaled W. Chehab
• [asterisk-users] Handle in dialplan user disconnection   Luiz Gustavo Chiaretto
• [asterisk-users] Handle in dialplan user disconnection   Luiz Gustavo Chiaretto
• [asterisk-users] Free calls to the US provider recommendation   Christian
• [asterisk-users] Debug Dropped Audio   Jesse Cloutier
• [asterisk-users] Upgrade and recompilation   Harel Cohen
• [asterisk-users] Zaptel slow to detect remote hangup   Tzafrir Cohen
• [asterisk-users] downgrade libpri   Tzafrir Cohen
• [asterisk-users] On-Hold Music   Tzafrir Cohen
• [asterisk-users] Hide the plain text password   Tzafrir Cohen
• [asterisk-users] Hide the plain text password   Tzafrir Cohen
• [asterisk-users] Hide the plain text password   Tzafrir Cohen
• [asterisk-users] On-Hold Music   Tzafrir Cohen
• [asterisk-users] Hide the plain text password (suggestion)   Tzafrir Cohen
• [asterisk-users] Hide the plain text password   Tzafrir Cohen
• [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?   Tzafrir Cohen
• [asterisk-users] Realtime and Local Channel Crash Problem 1.8.3-rc2   Nic Colledge
• [asterisk-users] Realtime and Local Channel Crash Problem 1.8.3-rc2   Nic Colledge
• [asterisk-users] Realtime and Local Channel Crash Problem 1.8.3-rc2   Nic Colledge
• [asterisk-users] Google 10%   Dean Collins
• [asterisk-users] Newbie´s question about Asterisk...   Dean Collins
• [asterisk-users] fail-over server   Carlos M Cruz
• [asterisk-users] Asterisk Performance   Gergo Csibra
• [asterisk-users] fail-over server   Gergo Csibra
• [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?   Ken D'Ambrosio
• [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?   Ken D'Ambrosio
• [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?   Ken D'Ambrosio
• [asterisk-users] Call files error   Tamás Dajka
• [asterisk-users] Call files error   Tamás Dajka
• [asterisk-users] Call files error   Tamás Dajka
• [asterisk-users] Call files error   Tamás Dajka
• [asterisk-users] Set variable on Call Answer   Dan Dan
• [asterisk-users] Set variable on Call Answer   Dan Dan
• [asterisk-users] Call Files, Variable passing   Dan Dan
• [asterisk-users] Voice mail forwarding enhancement   Matt Darnell
• [asterisk-users] IAX channel name incorrect - Found in 1.2 still happens in 1.6   Steve Davies
• [asterisk-users] Aastra phones cannot transfer calls?   Steve Davies
• [asterisk-users] Barge in.   Steve Davies
• [asterisk-users] IAX channel name incorrect - Found in 1.2 still happens in 1.6   Steve Davies
• [asterisk-users] Outgoing FXO calls have no audio with callprogress=no   Alec Davis
• [asterisk-users] Asterisk Performance   Juan David Diaz
• [asterisk-users] Setting two E1 cards   Juan David Diaz
• [asterisk-users] T1 PRI shows yellow/red alarm   Juan David Diaz
• [asterisk-users] T1 PRI shows yellow/red alarm   Juan David Diaz
• [asterisk-users] T1 PRI shows yellow/red alarm   Juan David Diaz
• [asterisk-users] T1 PRI shows yellow/red alarm   Juan David Diaz
• [asterisk-users] T1 PRI shows yellow/red alarm   Juan David Diaz
• [asterisk-users] Connecting to Cisco Iad2430 to Asterisk   Jim Dickenson
• [asterisk-users] Fax Woes   Mike Diehl
• [asterisk-users] Asterisk Call File using Local Channel not passing Variable back to Dialplan   Mike Diehl
• [asterisk-users] AGI script dies after receivefax   Mike Diehl
• [asterisk-users] AGI script dies after receivefax   Mike Diehl
• [asterisk-users] AGI script dies after receivefax   Mike Diehl
• [asterisk-users] Playback in uplink and recording in downlink   Felix Dong
• [asterisk-users] How to use Monitor() in Python AGI   Felix Dong
• [asterisk-users] Adjusting Rx and Tx gains   Felix Dong
• [asterisk-users] Adjusting Rx and Tx gains   Felix Dong
• [asterisk-users] how to diable echo cancellation for sip?   Felix Dong
• [asterisk-users] function Echo() doesn't work   Felix Dong
• [asterisk-users] function Echo() doesn't work   Felix Dong
• [asterisk-users] function Echo() doesn't work   Felix Dong
• [asterisk-users] function Echo() doesn't work   Felix Dong
• [asterisk-users] function Echo() doesn't work   Felix Dong
• [asterisk-users] function Echo() doesn't work   Felix Dong
• [asterisk-users] problems with voicemail and centos 5   Eric Doutreleau
• [asterisk-users] SIP friend name   Paul Dugas
• [asterisk-users] Inbound SIP calls work, just not when making calls between extensions.   Ernie Dunbar
• [asterisk-users] Defining what an extension should do after the Dial() command returns busy.   Ernie Dunbar
• [asterisk-users] Defining what an extension should do after the Dial() command returns busy.   Ernie Dunbar
• [asterisk-users] Defining what an extension should do after the Dial() command returns busy.   Ernie Dunbar
• [asterisk-users] Aastra phones cannot transfer calls?   Ernie Dunbar
• [asterisk-users] Aastra phones cannot transfer calls?   Ernie Dunbar
• [asterisk-users] Aastra phones cannot transfer calls?   Ernie Dunbar
• [asterisk-users] Aastra phones cannot transfer calls?   Ernie Dunbar
• [asterisk-users] Aastra phones cannot transfer calls?   Ernie Dunbar
• [asterisk-users] PRI B-Channel restarting itself continually   Ernie Dunbar
• [asterisk-users] fail-over server   Michelle Dupuis
• [asterisk-users] Voicemail email attachment as MP3, with tags containing sender name, number, message number   Michelle Dupuis
• [asterisk-users] Voicemail email attachment as MP3, with tags containing sender name, number, message number   Michelle Dupuis
• [asterisk-users] Multiple public address to one Asterisk server behind NAT?   Michelle Dupuis
• [asterisk-users] Multiple public address to one Asterisk serverbehind NAT?   Michelle Dupuis
• [asterisk-users] How to use Monitor() in Python AGI   Steve Edwards
• [asterisk-users] MP3 Crashing Asterisk   Steve Edwards
• [asterisk-users] Can a duration limit be specified in spool call file?   Steve Edwards
• [asterisk-users] Callback through extensions.conf?   Steve Edwards
• [asterisk-users] Callback through extensions.conf?   Steve Edwards
• [asterisk-users] queue called by agi doesn't re-enter the script   Steve Edwards
• [asterisk-users] Defining what an extension should do after the Dial() command returns busy.   Steve Edwards
• [asterisk-users] meetme conference & playback of random sound file   Steve Edwards
• [asterisk-users] dialplan announcements   Steve Edwards
• [asterisk-users] On-Hold Music   Steve Edwards
• [asterisk-users] On-Hold Music   Steve Edwards
• [asterisk-users] On-Hold Music   Steve Edwards
• [asterisk-users] IP ban list by country   Steve Edwards
• [asterisk-users] [1.4/AGI] CHANNEL STATUS never "down & available"?   Steve Edwards
• [asterisk-users] [1.4/AGI] CHANNEL STATUS never "down & available"?   Steve Edwards
• [asterisk-users] Newbie´s question about Asterisk...   Steve Edwards
• [asterisk-users] [1.4/AGI] CHANNEL STATUS never "down & available"?   Steve Edwards
• [asterisk-users] AGI script dies after receivefax   Steve Edwards
• [asterisk-users] [1.4] "show channels" in extensions.conf?   Steve Edwards
• [asterisk-users] calls between iax and sip   Steve Edwards
• [asterisk-users] [1.4.39.1/AGI] ast_carefulwrite: write() returned error: Broken pipe   Steve Edwards
• [asterisk-users] calls between iax and sip   Steve Edwards
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   Steve Edwards
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   Steve Edwards
• [asterisk-users] [1.4] Still can't get it to call back   Steve Edwards
• [asterisk-users] [1.4.39.2] Simple AGI doesn't reply   Steve Edwards
• [asterisk-users] missing argument on AGI   Steve Edwards
• [asterisk-users] Using voice modem as poor man's FXO in Asterisk 1.8   Steve Edwards
• [asterisk-users] Return variables from func_odbc calls?   Jose P. Espinal
• [asterisk-users] secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1   Jose P. Espinal
• [asterisk-users] secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1   Jose P. Espinal
• [asterisk-users] PRI voice optimization   C F
• [asterisk-users] Transfer Device Data   C F
• [asterisk-users] Hide the plain text password   C F
• [asterisk-users] Hide the plain text password   C F
• [asterisk-users] Problem installing FXS module in old digium 4 channel tdm card   C F
• [asterisk-users] Problem installing FXS module in old digium 4 channel tdm card   C F
• [asterisk-users] Problem installing FXS module in old digium 4 channel tdm card   C F
• [asterisk-users] Problem installing FXS module in old digium 4 channel tdm card   C F
• [asterisk-users] res_pgsql re-connect on db failure?   Bryan Field-Elliot
• [asterisk-users] cmd MySQL   Felipe Figueiredo
• [asterisk-users] Fwd: cmd MySQL   Felipe Figueiredo
• [asterisk-users] Fwd: cmd MySQL   Felipe Figueiredo
• [asterisk-users] On-Hold Music   Cary Fitch
• [asterisk-users] REFER and dialplan broken (as documentedinchan_sip.c on line 11951)   Cary Fitch
• [asterisk-users] REFER and dialplan broken(asdocumentedinchan_sip.c on line 11951)   Cary Fitch
• [asterisk-users] "Asterisk" caller ID   Cary Fitch
• [asterisk-users] Google Voice outbound Caller ID broken   Cary Fitch
• [asterisk-users] echo when calling to the pstn   Vitor Carlos Flausino
• [asterisk-users] sip trunk balancing   Kevin P. Fleming
• [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer   Kevin P. Fleming
• [asterisk-users] T.38 negotiation error   Kevin P. Fleming
• [asterisk-users] RTP keepalive doesn't work   Kevin P. Fleming
• [asterisk-users] Early audio SIP sequence order question   Kevin P. Fleming
• [asterisk-users] Early audio SIP sequence order question   Kevin P. Fleming
• [asterisk-users] digium te220   Kevin P. Fleming
• [asterisk-users] Fax for Asterisk SIP-TDM   Kevin P. Fleming
• [asterisk-users] [modules.conf] Modules still loaded after "noload"   Kevin P. Fleming
• [asterisk-users] Hide the plain text password   Kevin P. Fleming
• [asterisk-users] Hide the plain text password   Kevin P. Fleming
• [asterisk-users] Hide the plain text password   Kevin P. Fleming
• [asterisk-users] Fax Woes   Kevin P. Fleming
• [asterisk-users] Hide the plain text password   Kevin P. Fleming
• [asterisk-users] Hide the plain text password   Kevin P. Fleming
• [asterisk-users] further action after caller in a queue hangs up   Kevin P. Fleming
• [asterisk-users] Hide the plain text password   Kevin P. Fleming
• [asterisk-users] Polycom Do Not Disturb button and asterisk hints   Kevin P. Fleming
• [asterisk-users] SIP Hardphone that works well with asterisk   Kevin P. Fleming
• [asterisk-users] AGI script dies after receivefax   Kevin P. Fleming
• [asterisk-users] asterisk security....again   Kevin P. Fleming
• [asterisk-users] Asterisk 1.8.3   Vinícius Fontes
• [asterisk-users] Paging with Polycom 3.3.x   Dave Fullerton
• [asterisk-users] [OT] Yealink IP Phones   Jose Flores Galicia
• [asterisk-users] Question about EuroBRI final 2 digits   Christian Gansberger
• [asterisk-users] Gtalk/Jabber Issue   Jian Gao
• [asterisk-users] Hide the plain text password   Jian Gao
• [asterisk-users] Hide the plain text password   Jian Gao
• [asterisk-users] Hide the plain text password   Jian Gao
• [asterisk-users] Hide the plain text password   Jian Gao
• [asterisk-users] Playback in uplink and recording in downlink   Ruddy Gbaguidi
• [asterisk-users] duplicate keys change from zaptel to dahdi 2.4.0   Jerry Geis
• [asterisk-users] duplicate keys change from zaptel to dahdi 2.4.0   Jerry Geis
• [asterisk-users] Google Voice outbound Caller ID broken   Chris Gentle
• [asterisk-users] Google Voice outbound Caller ID broken   Chris Gentle
• [asterisk-users] [newbie] Conference call   Gilles
• [asterisk-users] [newbie] Conference call   Gilles
• [asterisk-users] Zaptel slow to detect remote hangup   Gilles
• [asterisk-users] Callback through extensions.conf?   Gilles
• [asterisk-users] Zaptel slow to detect remote hangup   Gilles
• [asterisk-users] Callback through extensions.conf?   Gilles
• [asterisk-users] Callback through extensions.conf?   Gilles
• [asterisk-users] Callback through extensions.conf?   Gilles
• [asterisk-users] Callback through extensions.conf?   Gilles
• [asterisk-users] Callback through extensions.conf?   Gilles
• [asterisk-users] Zaptel slow to detect remote hangup   Gilles
• [asterisk-users] Callback through extensions.conf?   Gilles
• [asterisk-users] Callback through extensions.conf?   Gilles
• [asterisk-users] Callback through extensions.conf?   Gilles
• [asterisk-users] Callback through extensions.conf?   Gilles
• [asterisk-users] Callback through extensions.conf?   Gilles
• [asterisk-users] Callback through extensions.conf?   Gilles
• [asterisk-users] Callback through extensions.conf?   Gilles
• [asterisk-users] Callback through extensions.conf?   Gilles
• [asterisk-users] Callback through extensions.conf?   Gilles
• [asterisk-users] Callback through extensions.conf?   Gilles
• [asterisk-users] Unable to make outgoing calls with Internode   Gilles
• [asterisk-users] [Zaptel] "numberplan-local" context from nowhere?   Gilles
• [asterisk-users] [Zaptel] "numberplan-local" context from nowhere?   Gilles
• [asterisk-users] [modules.conf] Modules still loaded after "noload"   Gilles
• [asterisk-users] [modules.conf] Modules still loaded after "noload"   Gilles
• [asterisk-users] [modules.conf] Modules still loaded after "noload"   Gilles
• [asterisk-users] [modules.conf] Modules still loaded after "noload"   Gilles
• [asterisk-users] Callback through extensions.conf?   Gilles
• [asterisk-users] Callback through extensions.conf?   Gilles
• [asterisk-users] Callback through extensions.conf?   Gilles
• [asterisk-users] Dial(Local/...) vs. Goto()?   Gilles
• [asterisk-users] Dial(Local/...) vs. Goto()?   Gilles
• [asterisk-users] [1.4/AGI] CHANNEL STATUS never "down & available"?   Gilles
• [asterisk-users] [1.4/AGI] CHANNEL STATUS never "down & available"?   Gilles
• [asterisk-users] [1.4/AGI] CHANNEL STATUS never "down & available"?   Gilles
• [asterisk-users] [1.4/AGI] CHANNEL STATUS never "down & available"?   Gilles
• [asterisk-users] [1.4] "show channels" in extensions.conf?   Gilles
• [asterisk-users] [1.4/AGI] CHANNEL STATUS never "down & available"?   Gilles
• [asterisk-users] [1.4] "show channels" in extensions.conf?   Gilles
• [asterisk-users] [Dahdi 2.4.0] DAHDI_CHANCONFIG failed on channel 1   Gilles
• [asterisk-users] [Dahdi 2.4.0] DAHDI_CHANCONFIG failed on channel 1   Gilles
• [asterisk-users] [1.4.39.1/AGI] ast_carefulwrite: write() returned error: Broken pipe   Gilles
• [asterisk-users] [1.4.39.1/AGI] ast_carefulwrite: write() returned error: Broken pipe   Gilles
• [asterisk-users] [1.4.39.1/AGI] ast_carefulwrite: write() returned error: Broken pipe   Gilles
• [asterisk-users] [1.4.39.2] Simple AGI doesn't reply   Gilles
• [asterisk-users] [1.4] Still can't get it to call back   Gilles
• [asterisk-users] [1.4.39.2] Simple AGI doesn't reply   Gilles
• [asterisk-users] [1.4.39.2] Simple AGI doesn't reply   Gilles
• [asterisk-users] [1.4] Still can't get it to call back   Gilles
• [asterisk-users] [1.4] Still can't get it to call back   Gilles
• [asterisk-users] [1.4] Still can't get it to call back   Gilles
• [asterisk-users] [1.4] Still can't get it to call back   Gilles
• [asterisk-users] [Dahdi 2.4.0] Flash() hangs up   Gilles
• [asterisk-users] Using voice modem as poor man's FXO in Asterisk 1.8   Gilles
• [asterisk-users] Obi110 as gateway to PSTN?   Gilles
• [asterisk-users] [Dahdi 2.4.0] Flash() hangs up   Gilles
• [asterisk-users] weird problem with Vega 100   Arie Goldfeld
• [asterisk-users] extend the timout on ringing for pri or sip   Israel Gottlieb
• [asterisk-users] extend the timout on ringing for pri or sip   Israel Gottlieb
• [asterisk-users] SIP Hardphone that works well with asterisk   Michael Graves
• [asterisk-users] Connect Asterisk to a cell phone   Michael Graves
• [asterisk-users] Connect Asterisk to a cell phone   Michael Graves
• [asterisk-users] Problems using Background within a macro on V 1.4   Paddy Grice
• [asterisk-users] how to get Current Calls details   Thorsten Göllner
• [asterisk-users] how to get Current Calls details   Thorsten Göllner
• [asterisk-users] voice quality measurement using dahdi_monitor   Thorsten Göllner
• [asterisk-users] PRI voice optimization   Thorsten Göllner
• [asterisk-users] Detect #,* DTMF in dialplan   Thorsten Göllner
• [asterisk-users] dialplan announcements   ERIC HERRON
• [asterisk-users] dialplan announcements   ERIC HERRON
• [asterisk-users] Polycom IP335   ERIC HERRON
• [asterisk-users] Polycom IP335   ERIC HERRON
• [asterisk-users] Polycom IP335   ERIC HERRON
• [asterisk-users] Polycom IP335   ERIC HERRON
• [asterisk-users] Polycom IP335   ERIC HERRON
• [asterisk-users] Polycom IP335   ERIC HERRON
• [asterisk-users] Polycom IP335   ERIC HERRON
• [asterisk-users] Polycom IP335   ERIC HERRON
• [asterisk-users] Got SIP response 400 "Bad Request" back from   ERIC HERRON
• [asterisk-users] zaptel/dahdi settings for singtel E1 line   Faisal Hanif
• [asterisk-users] dial option 'g' not working   Faisal Hanif
• [asterisk-users] Early audio SIP sequence order question   Faisal Hanif
• [asterisk-users] CDR with unix time.   Faisal Hanif
• [asterisk-users] Ported Asterisk in Android   Faisal Hanif
• [asterisk-users] Cisco 7960 & asterisk 1.8.22 ringlist.dat error   Faisal Hanif
• [asterisk-users] issue with some numbers   Faisal Hanif
• [asterisk-users] trunks and phones registered from the same IP   Faisal Hanif
• [asterisk-users] unregistered trunks and registered phones coming from the same IP   Faisal Hanif
• [asterisk-users] DTMF not detected, time out   Faisal Hanif
• [asterisk-users] Lua extensions are not working on asterisk 1.8.2.3   Faisal Hanif
• [asterisk-users] DTMF not detected, time out   Faisal Hanif
• [asterisk-users] DTMF not detected, time out   Faisal Hanif
• [asterisk-users] Connect Asterisk to a cell phone   Faisal Hanif
• [asterisk-users] How to know Caller's last position in Queue?   Faisal Hanif
• [asterisk-users] how to diable echo cancellation for sip?   Faisal Hanif
• [asterisk-users] function Echo() doesn't work   Faisal Hanif
• [asterisk-users] function Echo() doesn't work   Faisal Hanif
• [asterisk-users] function Echo() doesn't work   Faisal Hanif
• [asterisk-users] How to know Caller's last position in Queue?   Faisal Hanif
• [asterisk-users] function Echo() doesn't work   Faisal Hanif
• [asterisk-users] Play one audio file to the called part before the Dial() command‏   Faisal Hanif
• [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP   Faisal Hanif
• [asterisk-users] trunk not working if I register a phone at the same IP as the trunk peer's IP   Faisal Hanif
• [asterisk-users] pipe audio stream to external application   Faisal Hanif
• [asterisk-users] Asterisk on a USB with persistence   Faisal Hanif
• [asterisk-users] DTMF and Snom   Faisal Hanif
• [asterisk-users] Dial(Local/...) vs. Goto()?   Faisal Hanif
• [asterisk-users] Asterisk with TE 121 DADHI incoming calls fail   Faisal Hanif
• [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2   Faisal Hanif
• [asterisk-users] lua -asterisk manual   Faisal Hanif
• [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2   Faisal Hanif
• [asterisk-users] Fwd: cmd MySQL   Faisal Hanif
• [asterisk-users] DIAL through Specific number in PRI   Faisal Hanif
• [asterisk-users] AstLinux 0.7.6 Released   Darrick Hartman
• [asterisk-users] Barge in.   Peter den Hartog
• [asterisk-users] Barge in.   Peter den Hartog
• [asterisk-users] Two Asterisk machines for redundancy   Peter den Hartog
• [asterisk-users] SIP Hardphone that works well with asterisk   Gordon Henderson
• [asterisk-users] uptime   Gordon Henderson
• [asterisk-users] Asterisk 1.8.3 BLF stopped working   Rob Hillis
• [asterisk-users] Unknown calls   Rizwan Hisham
• [asterisk-users] extend the timout on ringing for pri or sip   Rizwan Hisham
• [asterisk-users] Unknown calls   Rizwan Hisham
• [asterisk-users] Problem in dialing out   Rizwan Hisham
• [asterisk-users] Unknown calls   Rizwan Hisham
• [asterisk-users] Carrying context from one server to another?   Rizwan Hisham
• [asterisk-users] asterisk security....again   Rizwan Hisham
• [asterisk-users] asterisk security....again   Rizwan Hisham
• [asterisk-users] asterisk security....again   Rizwan Hisham
• [asterisk-users] asterisk security....again   Rizwan Hisham
• [asterisk-users] asterisk security....again   Rizwan Hisham
• [asterisk-users] T1 PRI shows yellow/red alarm   Dean Hoover
• [asterisk-users] T1 PRI shows yellow/red alarm   Dean Hoover
• [asterisk-users] T1 PRI shows yellow/red alarm   Dean Hoover
• [asterisk-users] T1 PRI shows yellow/red alarm   Dean Hoover
• [asterisk-users] ${HANGUPCAUSE} in CDR   Steve Howes
• [asterisk-users] forward calls by the ports   Steve Howes
• [asterisk-users] On-Hold Music   Steve Howes
• [asterisk-users] Hide the plain text password   Steve Howes
• [asterisk-users] uptime   Steve Howes
• [asterisk-users] Hide the plain text password   Steve Howes
• [asterisk-users] Asterisk Using as a SIP Client   Steven Howes
• [asterisk-users] Regarding Asterisk   Steven Howes
• [asterisk-users] asterisk security....again   Steven Howes
• [asterisk-users] PRI voice optimization   DHAVAL INDRODIYA
• [asterisk-users] voice quality measurement using dahdi_monitor   DHAVAL INDRODIYA
• [asterisk-users] PRI voice optimization   DHAVAL INDRODIYA
• [asterisk-users] Newbie´s question about Asterisk...   DHAVAL INDRODIYA
• [asterisk-users] Meet me recording   DHAVAL INDRODIYA
• [asterisk-users] DIAL through Specific number in PRI   DHAVAL INDRODIYA
• [asterisk-users] Aastra phones cannot transfer calls?   Ira
• [asterisk-users] 1.8.2.4: SIP dialogs not killed?   Johannes Jakob
• [asterisk-users] 1.8.2.4: SIP dialogs not killed?   Johannes Jakob
• [asterisk-users] 1.8.2.4: SIP dialogs not killed?   Johannes Jakob
• [asterisk-users] How to disable srtp in asterisk 1.8.2.3?   Kai-Uwe Jensen
• [asterisk-users] SIP Hardphone that works well with asterisk   Kai-Uwe Jensen
• [asterisk-users] SIP Hardphone that works well with asterisk   Andrew Joakimsen
• [asterisk-users] alarm POTS lines   Andrew Joakimsen
• [asterisk-users] TLS/SRTP calls go to circuit busy.   mitch Johnson
• [asterisk-users] One way dialing over a SIP trunk   Mitch Johnson
• [asterisk-users] Question about how traffic passes from phones   Mitch Johnson
• [asterisk-users] meetme conference & playback of random sound file   John Jolly
• [asterisk-users] DTMF and Snom   Dan Journo
• [asterisk-users] Asterisk Performance   Arstan Jusupov
• [asterisk-users] Gtalk/Jabber Issue   Arstan Jusupov
• [asterisk-users] Musiconhold priority   Jonas Kellens
• [asterisk-users] How to load new musiconhold classes ?   Jonas Kellens
• [asterisk-users] Realtime queues not playing prompts   Jonas Kellens
• [asterisk-users] Realtime queues not playing prompts   Jonas Kellens
• [asterisk-users] DTMF and Snom   Jonas Kellens
• [asterisk-users] Difference mohsuggest & mohinterpret   Jonas Kellens
• [asterisk-users] Hide the plain text password   Richard Kenner
• [asterisk-users] Hide the plain text password   Richard Kenner
• [asterisk-users] Hide the plain text password   Richard Kenner
• [asterisk-users] Hide the plain text password   Richard Kenner
• [asterisk-users] Hide the plain text password   Richard Kenner
• [asterisk-users] Hide the plain text password   Richard Kenner
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   Richard Kenner
• [asterisk-users] IP ban list by country   Bill Kenworthy
• [asterisk-users] CDR with unix time.   Mindaugas Kezys
• [asterisk-users] Play one audio file to the called part before the Dial() command‏   Mohammad Khan
• [asterisk-users] ${HANGUPCAUSE} in CDR   Shariq Khan
• [asterisk-users] meetme conference & playback of random sound file   John Kiniston
• [asterisk-users] Multiple public address to one Asterisk server behind NAT?   John Kiniston
• [asterisk-users] (no subject)   Kevin Kirts
• [asterisk-users] SoftHangup on asterisk 1.8.2.3   Jeremy Kister
• [asterisk-users] Hide the plain text password   Jeremy Kister
• [asterisk-users] Adhearsion 1.0.1 Released   Ben Klang
• [asterisk-users] missing argument on AGI   Ben Klang
• [asterisk-users] Upgrade and recompilation   Barry L. Kline
• [asterisk-users] My new blog http://cciev.ciscovoicetech.com/   Arun Kumar
• [asterisk-users] uptime   Jeff LaCoursiere
• [asterisk-users] uptime   Jeff LaCoursiere
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   Jeff LaCoursiere
• [asterisk-users] [newbie] Conference call   Pezhman Lali
• [asterisk-users] Callback through extensions.conf?   Pezhman Lali
• [asterisk-users] secure sccp   Pezhman Lali
• [asterisk-users] Using files .call or AMI   Pezhman Lali
• [asterisk-users] Fax for Asterisk SIP-TDM   Pezhman Lali
• [asterisk-users] trunks and phones registered from the same IP   Pezhman Lali
• [asterisk-users] further action after caller in a queue hangs up   Pezhman Lali
• [asterisk-users] unregistered trunks and registered phones coming from the same IP   Pezhman Lali
• [asterisk-users] Dial command   Pezhman Lali
• [asterisk-users] changing logo of 7905   Pezhman Lali
• [asterisk-users] Cisco 7945G phone with asterisk   Pezhman Lali
• [asterisk-users] DTMF not detected, time out   Pezhman Lali
• [asterisk-users] Two Asterisk machines for redundancy   Pezhman Lali
• [asterisk-users] Two Asterisk machines for redundancy   Pezhman Lali
• [asterisk-users] fail-over server   Edwin Lam
• [asterisk-users] CDR with unix time.   Rodrigo Lang
• [asterisk-users] CDR with unix time.   Rodrigo Lang
• [asterisk-users] CDR with unix time.   Rodrigo Lang
• [asterisk-users] Realtime MySQL - Asterisk 1.8.2   Rodrigo Lang
• [asterisk-users] Realtime MySQL - Asterisk 1.8.2   Rodrigo Lang
• [asterisk-users] extensions.lua with luasql.mysql.   Rodrigo Lang
• [asterisk-users] extensions.lua with luasql.mysql.   Rodrigo Lang
• [asterisk-users] SIP Hardphone that works well with asterisk   Andrew Latham
• [asterisk-users] SIP Hardphone that works well with asterisk   Andrew Latham
• [asterisk-users] SIP Hardphone that works well with asterisk   Andrew Latham
• [asterisk-users] Possible dumb question: new kernel, new DAHDI?   Andrew Latham
• [asterisk-users] SIP Hardphone that works well with asterisk   Andrew Latham
• [asterisk-users] Hide the plain text password   Andrew Latham
• [asterisk-users] Cisco 7945G phone with asterisk   Andrew Latham
• [asterisk-users] Connect Asterisk to a cell phone   Andrew Latham
• [asterisk-users] Polycom IP335   Andrew Latham
• [asterisk-users] Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available   Andrew Latham
• [asterisk-users] Recieve_Fax caused crash 1.8.2.3   Andrew Latham
• [asterisk-users] Recieve_Fax caused crash 1.8.2.3   Andrew Latham
• [asterisk-users] Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available   Andrew Latham
• [asterisk-users] Recieve_Fax caused crash 1.8.2.3   Andrew Latham
• [asterisk-users] exceeds the maximum size of ast_fdset error on Asterisk-1.8.0   Tilghman Lesher
• [asterisk-users] Return variables from func_odbc calls?   Tilghman Lesher
• [asterisk-users] Return variables from func_odbc calls?   Tilghman Lesher
• [asterisk-users] AGI script exits non-zero when running system command   Tilghman Lesher
• [asterisk-users] asterisk18 rpm issues   Tilghman Lesher
• [asterisk-users] Zaptel slow to detect remote hangup   Tilghman Lesher
• [asterisk-users] Call Recording audio file quality query   Tilghman Lesher
• [asterisk-users] Call Recording audio file quality query   Tilghman Lesher
• [asterisk-users] Callback through extensions.conf?   Tilghman Lesher
• [asterisk-users] AEL Eswitches   Tilghman Lesher
• [asterisk-users] CDR with unix time.   Tilghman Lesher
• [asterisk-users] CDR with unix time.   Tilghman Lesher
• [asterisk-users] CDR with unix time.   Tilghman Lesher
• [asterisk-users] On-Hold Music   Tilghman Lesher
• [asterisk-users] On-Hold Music   Tilghman Lesher
• [asterisk-users] Lua extensions are not working on asterisk 1.8.2.3   Tilghman Lesher
• [asterisk-users] uptime   Tilghman Lesher
• [asterisk-users] application for voice modulation   Tilghman Lesher
• [asterisk-users] pbx_ael.so: undefined symbol: ast_compile_ael2   Tilghman Lesher
• [asterisk-users] NVFaxDetect causing segfault   Tilghman Lesher
• [asterisk-users] [1.4] Still can't get it to call back   Tilghman Lesher
• [asterisk-users] Weird Inbound Problem.   iptel.co Lists
• [asterisk-users] Polycom IP335   Patrick Lists
• [asterisk-users] Polycom IP335   Patrick Lists
• [asterisk-users] Polycom IP335   Patrick Lists
• [asterisk-users] Polycom IP335   Patrick Lists
• [asterisk-users] Polycom IP335   Patrick Lists
• [asterisk-users] Polycom IP335   Patrick Lists
• [asterisk-users] Polycom IP335   Patrick Lists
• [asterisk-users] Polycom IP335   Patrick Lists
• [asterisk-users] Polycom IP335   Patrick Lists
• [asterisk-users] Regarding Asterisk   Patrick Lists
• [asterisk-users] asterisk18 rpm issues   Frank Liu
• [asterisk-users] asterisk18 rpm issues   Frank Liu
• [asterisk-users] echo when calling to the pstn   Ye Liu
• [asterisk-users] Using voice modem as poor man's FXO in Asterisk 1.8   Stuart Longland
• [asterisk-users] Using voice modem as poor man's FXO in Asterisk 1.8   Stuart Longland
• [asterisk-users] Dialplan end of pattern matching question   Gabriel Ortiz Lour
• [asterisk-users] Samsung smt-i3100   Julian Lyndon-Smith
• [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer   Doug Lytle
• [asterisk-users] dialplan announcements   Doug Lytle
• [asterisk-users] Paging a message. How?   Doug Lytle
• [asterisk-users] Voicemail email attachment as MP3, with tags containing sender name, number, message number   Doug Lytle
• [asterisk-users] Dialplan end of pattern matching question   Doug Lytle
• [asterisk-users] Handle in dialplan user disconnection   Doug Lytle
• [asterisk-users] Handle in dialplan user disconnection   Doug Lytle
• [asterisk-users] Asterisk Performance   Leif Madsen
• [asterisk-users] [modules.conf] Modules still loaded after "noload"   Leif Madsen
• [asterisk-users] Hide the plain text password   Leif Madsen
• [asterisk-users] Hide the plain text password   Leif Madsen
• [asterisk-users] Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available   Leif Madsen
• [asterisk-users] secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1   Leif Madsen
• [asterisk-users] secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1   Leif Madsen
• [asterisk-users] Recieve_Fax caused crash 1.8.2.3   Leif Madsen
• [asterisk-users] Recieve_Fax caused crash 1.8.2.3   Leif Madsen
• [asterisk-users] how to get Current Calls details   Ishfaq Malik
• [asterisk-users] how to get Current Calls details   Ishfaq Malik
• [asterisk-users] MeetMe and admin users   Ishfaq Malik
• [asterisk-users] MeetMe and admin users   Ishfaq Malik
• [asterisk-users] About maxlen parameter in queues   Ishfaq Malik
• [asterisk-users] Call Recording audio file quality query   Ishfaq Malik
• [asterisk-users] Call Recording audio file quality query   Ishfaq Malik
• [asterisk-users] Reliably getting sip extension name from channel variables   Ishfaq Malik
• [asterisk-users] Asterisk 1.8.3   Ishfaq Malik
• [asterisk-users] Asterisk 1.8.3   Ishfaq Malik
• [asterisk-users] Asterisk 1.8.3   Ishfaq Malik
• [asterisk-users] Realtime MySQL - Asterisk 1.8.2   Ishfaq Malik
• [asterisk-users] SIP Hardphone that works well with asterisk   Ishfaq Malik
• [asterisk-users] Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available   Ishfaq Malik
• [asterisk-users] AMI FullyBooted issue   Ishfaq Malik
• [asterisk-users] AMI FullyBooted issue   Ishfaq Malik
• [asterisk-users] Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available   Ishfaq Malik
• [asterisk-users] Asterisk 1.8.3-rc3 and one way audio   Ishfaq Malik
• [asterisk-users] AEL Eswitches   Thiago Maluf
• [asterisk-users] How to know Caller's last position in Queue?   Asterisk Man
• [asterisk-users] How to know Caller's last position in Queue?   Asterisk Man
• [asterisk-users] Two Asterisk machines for redundancy   Nicolás Aldo Martínez
• [asterisk-users] IP ban list by country   John Marvin
• [asterisk-users] SIP Hardphone that works well with asterisk   Joel Maslak
• [asterisk-users] Callback through extensions.conf?   Sherwood McGowan
• [asterisk-users] Any voice changer applications for Asterisk?   Sherwood McGowan
• [asterisk-users] Callback through extensions.conf?   Sherwood McGowan
• [asterisk-users] Callback through extensions.conf?   Sherwood McGowan
• [asterisk-users] Can a duration limit be specified in spool call file?   Sherwood McGowan
• [asterisk-users] Callback through extensions.conf?   Sherwood McGowan
• [asterisk-users] Callback through extensions.conf?   Sherwood McGowan
• [asterisk-users] Error: Unable to create channel of type 'SIP'   Sherwood McGowan
• [asterisk-users] Callback through extensions.conf?   Sherwood McGowan
• [asterisk-users] Can a duration limit be specified in spool call file?   Sherwood McGowan
• [asterisk-users] Can a duration limit be specified in spool call file?   Sherwood McGowan
• [asterisk-users] remote bridging   Sherwood McGowan
• [asterisk-users] Callback through extensions.conf?   Sherwood McGowan
• [asterisk-users] Can a duration limit be specified in spool call file?   Sherwood McGowan
• [asterisk-users] Call Recording audio file quality query   Sherwood McGowan
• [asterisk-users] Call Recording audio file quality query   Sherwood McGowan
• [asterisk-users] Call Recording audio file quality query   Sherwood McGowan
• [asterisk-users] Call Recording audio file quality query   Sherwood McGowan
• [asterisk-users] Call files error   Sherwood McGowan
• [asterisk-users] Set variable on Call Answer   Sherwood McGowan
• [asterisk-users] Callback through extensions.conf?   Sherwood McGowan
• [asterisk-users] Call Recording audio file quality query   Sherwood McGowan
• [asterisk-users] Call Recording audio file quality query   Sherwood McGowan
• [asterisk-users] Variables losing their value????   Sherwood McGowan
• [asterisk-users] Variables losing their value????   Sherwood McGowan
• [asterisk-users] SIP Hardphone that works well with asterisk   Sherwood McGowan
• [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer   Mike
• [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer   Mike
• [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer   Mike
• [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer   Mike
• [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer   Mike
• [asterisk-users] Polycom Do Not Disturb button and asterisk hints   Mike
• [asterisk-users] Paging with Polycom 3.3.x   Mike
• [asterisk-users] Paging with Polycom 3.3.x   Mike
• [asterisk-users] Paging with Polycom 3.3.x   Mike
• [asterisk-users] Paging with Polycom 3.3.x   Mike
• [asterisk-users] Paging with Polycom 3.3.x   Mike
• [asterisk-users] Gtalk/Jabber Issue   Vladimir Mikhelson
• [asterisk-users] Gtalk/Jabber Issue   Vladimir Mikhelson
• [asterisk-users] Gtalk/Jabber Issue   Vladimir Mikhelson
• [asterisk-users] Gtalk/Jabber Issue   Vladimir Mikhelson
• [asterisk-users] Gtalk/Jabber Issue   Vladimir Mikhelson
• [asterisk-users] Gtalk/Jabber Issue   Vladimir Mikhelson
• [asterisk-users] Gtalk/Jabber Issue   Vladimir Mikhelson
• [asterisk-users] Gtalk/Jabber Issue   Vladimir Mikhelson
• [asterisk-users] Google Voice outbound Caller ID broken   Vladimir Mikhelson
• [asterisk-users] Google Voice outbound Caller ID broken   Vladimir Mikhelson
• [asterisk-users] Google Voice outbound Caller ID broken   Vladimir Mikhelson
• [asterisk-users] uptime   Barry Miller
• [asterisk-users] Cisco 7960 & asterisk 1.8.22 ringlist.dat error   James Miller
• [asterisk-users] Cisco 7960 & asterisk 1.8.22 ringlist.dat error   James Miller
• [asterisk-users] Cisco 7960 & asterisk 1.8.22 ringlist.dat error   James Miller
• [asterisk-users] Cisco 7960 & asterisk 1.8.22 ringlist.dat error   James Miller
• [asterisk-users] Cisco 7960 & asterisk 1.8.22 ringlist.dat error   James Miller
• [asterisk-users] Cisco 7960 & asterisk 1.8.22 ringlist.dat error   James Miller
• [asterisk-users] AstMail   Flavio Miranda
• [asterisk-users] terrible MeetMe sound with 1.6.2.9   Louis-David Mitterrand
• [asterisk-users] terrible MeetMe sound with 1.6.2.9   Louis-David Mitterrand
• [asterisk-users] terrible MeetMe sound with 1.6.2.9   Louis-David Mitterrand
• [asterisk-users] MeetMe and admin users   Tony Mountifield
• [asterisk-users] [1.4] Still can't get it to call back   Tony Mountifield
• [asterisk-users] [1.4] Still can't get it to call back   Tony Mountifield
• [asterisk-users] [1.4] Still can't get it to call back   Tony Mountifield
• [asterisk-users] SIP Hardphone that works well with asterisk   Thomas Mullins
• [asterisk-users] Error: Unable to create channel of type 'SIP'   RSCL Mumbai
• [asterisk-users] Transfer Device Data   Elliot Murdock
• [asterisk-users] SIP session timers just on one specific channel   Guido Negro
• [asterisk-users] SIP session timers just on one specific channel   Guido Negro
• [asterisk-users] Musiconhold priority   Danny Nicholas
• [asterisk-users] How to load new musiconhold classes ?   Danny Nicholas
• [asterisk-users] How to disable srtp in asterisk 1.8.2.3?   Danny Nicholas
• [asterisk-users] How to disable srtp in asterisk 1.8.2.3?   Danny Nicholas
• [asterisk-users] Outgoing agent´s calls   Danny Nicholas
• [asterisk-users] Outgoing agent´s calls   Danny Nicholas
• [asterisk-users] About maxlen parameter in queues   Danny Nicholas
• [asterisk-users] About maxlen parameter in queues   Danny Nicholas
• [asterisk-users] Callback through extensions.conf?   Danny Nicholas
• [asterisk-users] OT: SwitchVox Mailing List?   Danny Nicholas
• [asterisk-users] multiple inbound calls from same sip trunk   Danny Nicholas
• [asterisk-users] terrible MeetMe sound with 1.6.2.9   Danny Nicholas
• [asterisk-users] Defining what an extension should do after the Dial() command returns busy.   Danny Nicholas
• [asterisk-users] Question about EuroBRI final 2 digits   Danny Nicholas
• [asterisk-users] Asterisk 1.8.3   Danny Nicholas
• [asterisk-users] On-Hold Music   Danny Nicholas
• [asterisk-users] On-Hold Music   Danny Nicholas
• [asterisk-users] issue with some numbers   Danny Nicholas
• [asterisk-users] issue with some numbers   Danny Nicholas
• [asterisk-users] issue with some numbers   Danny Nicholas
• [asterisk-users] On-Hold Music   Danny Nicholas
• [asterisk-users] Hide the plain text password   Danny Nicholas
• [asterisk-users] On-Hold Music   Danny Nicholas
• [asterisk-users] On-Hold Music   Danny Nicholas
• [asterisk-users] On-Hold Music   Danny Nicholas
• [asterisk-users] Hide the plain text password   Danny Nicholas
• [asterisk-users] Adjusting Rx and Tx gains   Danny Nicholas
• [asterisk-users] further action after caller in a queue hangs up   Danny Nicholas
• [asterisk-users] Paging a message. How?   Danny Nicholas
• [asterisk-users] Voicemail email attachment as MP3, with tags containing sender name, number, message number   Danny Nicholas
• [asterisk-users] Dialplan end of pattern matching question   Danny Nicholas
• [asterisk-users] No ring tone on inbound call - but channelconnects fine   Danny Nicholas
• [asterisk-users] calls are not going thru e1 line   Danny Nicholas
• [asterisk-users] Regarding Asterisk   Danny Nicholas
• [asterisk-users] Pickup from an specific exten   Danny Nicholas
• [asterisk-users] Assigning an extension to a roaming phone   Danny Nicholas
• [asterisk-users] Meet me recording   Danny Nicholas
• [asterisk-users] Assigning an extension to a roaming phone   Danny Nicholas
• [asterisk-users] Assigning an extension to a roaming phone   Danny Nicholas
• [asterisk-users] Assigning an extension to a roaming phone   Danny Nicholas
• [asterisk-users] Assigning an extension to a roaming phone   Danny Nicholas
• [asterisk-users] Assigning an extension to a roaming phone   Danny Nicholas
• [asterisk-users] Assigning an extension to a roaming phone   Danny Nicholas
• [asterisk-users] calls between iax and sip   Danny Nicholas
• [asterisk-users] Multiple public address to one Asterisk serverbehind NAT?   Danny Nicholas
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   Danny Nicholas
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   Danny Nicholas
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   Danny Nicholas
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   Danny Nicholas
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   Danny Nicholas
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   Danny Nicholas
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   Danny Nicholas
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   Danny Nicholas
• [asterisk-users] REFER and dialplan broken (asdocumentedinchan_sip.c on line 11951)   Danny Nicholas
• [asterisk-users] missing argument on AGI   Danny Nicholas
• [asterisk-users] Asterisk/Skype   Danny Nicholas
• [asterisk-users] 1.8.2.4: SIP dialogs not killed?   Danny Nicholas
• [asterisk-users] Failover Routing   Deepika Nijhawan
• [asterisk-users] how to get Current Calls details   Nikhil
• [asterisk-users] how to get Current Calls details   Nikhil
• [asterisk-users] how to get Current Calls details   Nikhil
• [asterisk-users] how to get Current Calls details   Nikhil
• [asterisk-users] Radius Based Accounting for Asterisk   Nikhil
• [asterisk-users] Ported Asterisk in Android   Nikhil
• [asterisk-users] Ported Asterisk in Android   Nikhil
• [asterisk-users] Asterisk Using as a SIP Client   Nikhil
• [asterisk-users] Asterisk Using as a SIP Client   Nikhil
• [asterisk-users] Callback through extensions.conf?   John Novack
• [asterisk-users] Can a duration limit be specified in spool call file?   John Novack
• [asterisk-users] Call Files, Variable passing   Neal O'Mara
• [asterisk-users] Setting two E1 cards   Alejandro Cabrera Obed
• [asterisk-users] Newbie´s question about Asterisk...   Francisco Javier Cintrón Olguín
• [asterisk-users] Newbie´s question about Asterisk...   Francisco Javier Cintrón Olguín
• [asterisk-users] Newbie´s question about Asterisk...   Francisco Javier Cintrón Olguín
• [asterisk-users] Newbie´s question about Asterisk...   Francisco Javier Cintrón Olguín
• [asterisk-users] Handle in dialplan user disconnection   Sidarta Aguiar de Oliveira
• [asterisk-users] B410P: DAHDI BRI PTMP HDLC Abort (6) on Primary D-channel of span 1 (TEI Errors)   Olivier
• [asterisk-users] Polycom Blf / Directed Pickup   Olivier
• [asterisk-users] Question about EuroBRI final 2 digits   Olivier
• [asterisk-users] voice quality measurement using dahdi_monitor   Sevana Oy
• [asterisk-users] AstriEurope coference   Sevana Oy
• [asterisk-users] Early audio SIP sequence order question   Benoit Panizzon
• [asterisk-users] Early audio SIP sequence order question   Benoit Panizzon
• [asterisk-users] Early audio SIP sequence order question   Benoit Panizzon
• [asterisk-users] asterisk18 rpm issues   Jason Parker
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   Jason Parker
• [asterisk-users] no progress indication   Satish Patel
• [asterisk-users] Meet me recording   Satish Patel
• [asterisk-users] Unknown calls   Satish Patel
• [asterisk-users] Failover Routing   Satish Patel
• [asterisk-users] Question about EuroBRI final 2 digits   Roberto Piola
• [asterisk-users] Lua extensions are not working on asterisk 1.8.2.3   Carlo Pires
• [asterisk-users] Lua extensions are not working on asterisk 1.8.2.3   Carlo Pires
• [asterisk-users] Lua extensions are not working on asterisk 1.8.2.3   Carlo Pires
• [asterisk-users] Hide the plain text password   Dave Platt
• [asterisk-users] Regarding Asterisk   Fred Posner
• [asterisk-users] Using files .call or AMI   Edwin Quijada
• [asterisk-users] Using files .call or AMI   Edwin Quijada
• [asterisk-users] Using files .call or AMI   Edwin Quijada
• [asterisk-users] Using files .call or AMI   Edwin Quijada
• [asterisk-users] Using files .call or AMI   Edwin Quijada
• [asterisk-users] Using a Virtual IP Line   Edwin Quijada
• [asterisk-users] Microsoft Speech Server/UCMA Integration   RR
• [asterisk-users] NVFaxDetect causing segfault   Shamus Rask
• [asterisk-users] Dialplan execution stops on app call even with TryExec (Am I missing something simple?)   Jay Reeder
• [asterisk-users] Dialplan execution stops on app call even with TryExec (Am I missing something simple?)   Jay Reeder
• [asterisk-users] SIP Hardphone that works well with asterisk   Matt Riddell
• [asterisk-users] B410P: DAHDI BRI PTMP HDLC Abort (6) on Primary D-channel of span 1 (TEI Errors)   Matt Riddell
• [asterisk-users] Gtalk/Jabber Issue   Matt Riddell
• [asterisk-users] Trunk grouping   Malvin Rito
• [asterisk-users] Two Asterisk machines for redundancy   Aaron Roberts
• [asterisk-users] Unable to make outgoing calls with Internode   Da Rock
• [asterisk-users] Unable to make outgoing calls with Internode   Da Rock
• [asterisk-users] missing argument on AGI   Ron
• [asterisk-users] missing argument on AGI   Ron
• [asterisk-users] [Dahdi 2.4.0] DAHDI_CHANCONFIG failed on channel 1   Shaun Ruffell
• [asterisk-users] Problem installing FXS module in old digium 4 channel tdm card   Shaun Ruffell
• [asterisk-users] Problem installing FXS module in old digium 4 channel tdm card   Shaun Ruffell
• [asterisk-users] Problems with realtime sip   Shaymardanov Rushan
• [asterisk-users] Problems with realtime sip   Shaymardanov Rushan
• [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer   Tom Rymes
• [asterisk-users] Callback through extensions.conf?   Tom Rymes
• [asterisk-users] Cisco 7960 & asterisk 1.8.22 ringlist.dat error   Tom Rymes
• [asterisk-users] Trunk grouping   Damian Ryszka
• [asterisk-users] dial option 'g' not working   M S
• [asterisk-users] end a call after a specific time period   ABBAS SHAKEEL
• [asterisk-users] Detect DTMF tone during call?   Dan Saul
• [asterisk-users] Detect DTMF tone during call?   Dan Saul
• [asterisk-users] Free calls to the US provider recommendation   C. Savinovich
• [asterisk-users] Busy Detection on Analog Lines   Sebastian
• [asterisk-users] Musiconhold priority   Warren Selby
• [asterisk-users] Callback through extensions.conf?   Warren Selby
• [asterisk-users] Inbound SIP calls work, just not when making calls between extensions.   Warren Selby
• [asterisk-users] terrible MeetMe sound with 1.6.2.9   Warren Selby
• [asterisk-users] Gtalk/Jabber Issue   Warren Selby
• [asterisk-users] Gtalk/Jabber Issue   Warren Selby
• [asterisk-users] On-Hold Music   Warren Selby
• [asterisk-users] Voicemail email attachment as MP3, with tags containing sender name, number, message number   Warren Selby
• [asterisk-users] Dialplan end of pattern matching question   Warren Selby
• [asterisk-users] MEMBERINTERFACE and MEMBERNAME questions   Warren Selby
• [asterisk-users] Assigning an extension to a roaming phone   Warren Selby
• [asterisk-users] Assigning an extension to a roaming phone   Warren Selby
• [asterisk-users] [1.4.39.1/AGI] ast_carefulwrite: write() returned error: Broken pipe   Warren Selby
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   Warren Selby
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   Warren Selby
• [asterisk-users] DIAL through Specific number in PRI   Warren Selby
• [asterisk-users] extend the timout on ringing for pri or sip   Warren Selby
• [asterisk-users] Registration failed though configured.   Warren Selby
• [asterisk-users] Assigning an extension to a roaming phone   Warren Selby
• [asterisk-users] Debug Dropped Audio   Warren Selby
• [asterisk-users] Registration failed though configured.   Warren Selby
• [asterisk-users] CEL and PGSQL   Warren Selby
• [asterisk-users] Question about how traffic passes from phones   Warren Selby
• [asterisk-users] [OT] Yealink IP Phones   César Sequeira
• [asterisk-users] multiple inbound calls from same sip trunk   Mohan Shahi
• [asterisk-users] sangoma wanpipe install error   Moises Silva
• [asterisk-users] sangoma wanpipe install error   Moises Silva
• [asterisk-users] FAX on PRI to MFCR2   Moises Silva
• [asterisk-users] Question about EuroBRI final 2 digits   Cassius Smith
• [asterisk-users] Question about EuroBRI final 2 digits   Cassius Smith
• [asterisk-users] no progress indication   Cassius Smith
• [asterisk-users] no progress indication   Cassius Smith
• [asterisk-users] Need to buy the Digium card, to confirm   Cassius Smith
• [asterisk-users] MP3 Crashing Asterisk   Timothy Smith
• [asterisk-users] MP3 Crashing Asterisk   Timothy Smith
• [asterisk-users] MP3 Crashing Asterisk   Timothy Smith
• [asterisk-users] AGI script exits non-zero when running system command   Charles Solar
• [asterisk-users] AGI script exits non-zero when running system command   Charles Solar
• [asterisk-users] asterisk security....again   Jamie A. Stapleton
• [asterisk-users] MP3 Crashing Asterisk   A J Stiles
• [asterisk-users] MP3 Crashing Asterisk   A J Stiles
• [asterisk-users] MP3 Crashing Asterisk   A J Stiles
• [asterisk-users] Possible dumb question: new kernel, new DAHDI?   A J Stiles
• [asterisk-users] uptime   A J Stiles
• [asterisk-users] Hide the plain text password   A J Stiles
• [asterisk-users] Using a Virtual IP Line   A J Stiles
• [asterisk-users] asterisk security....again   A J Stiles
• [asterisk-users] PRI voice optimization   William Stillwell
• [asterisk-users] OT: SwitchVox Mailing List?   William Stillwell
• [asterisk-users] Call Recording audio file quality query   William Stillwell
• [asterisk-users] Gtalk/Jabber Issue   William Stillwell
• [asterisk-users] Gtalk/Jabber Issue   William Stillwell
• [asterisk-users] Gtalk/Jabber Issue   William Stillwell
• [asterisk-users] Gtalk/Jabber Issue   William Stillwell
• [asterisk-users] Gtalk/Jabber Issue   William Stillwell
• [asterisk-users] Gtalk/Jabber Issue   William Stillwell
• [asterisk-users] Gtalk/Jabber Issue   William Stillwell
• [asterisk-users] Gtalk/Jabber Issue   William Stillwell
• [asterisk-users] Gtalk/Jabber Issue   William Stillwell
• [asterisk-users] uptime   William Stillwell
• [asterisk-users] Gtalk/Jabber Issue   William Stillwell
• [asterisk-users] Gtalk/Jabber Issue   William Stillwell
• [asterisk-users] Gtalk/Jabber Issue   William Stillwell
• [asterisk-users] Google Voice outbound Caller ID broken   William Stillwell
• [asterisk-users] Paging with Polycom 3.3.x   William Stillwell
• [asterisk-users] Asterisk/Skype   William Stillwell
• [asterisk-users] Asterisk/Skype   William Stillwell
• [asterisk-users] zaptel/dahdi settings for singtel E1 line   Roi Stork
• [asterisk-users] sangoma wanpipe install error   Roi Stork
• [asterisk-users] sangoma wanpipe install error   Roi Stork
• [asterisk-users] "intercom" SIP header being ignored by Kirk wireless handsets   SIP Support
• [asterisk-users] AMI FullyBooted issue   Marcin Szymański
• [asterisk-users] issue with some numbers   Administrator TOOTAI
• [asterisk-users] Got SIP response 400 "Bad Request" back from   Christian Tardif
• [asterisk-users] Scheduled Maintenance: wiki.asterisk.org and code.asterisk.org   Asterisk Development Team
• [asterisk-users] Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.4.40 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.6.2.17 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 1.8.3 Now Available   Asterisk Development Team
• [asterisk-users] AST-2011-002: Multiple array overflow and crash vulnerabilities in UDPTL code   Asterisk Security Team
• [asterisk-users] Question about EuroBRI final 2 digits   Andrew Thomas
• [asterisk-users] Question about EuroBRI final 2 digits   Andrew Thomas
• [asterisk-users] calls are not going thru e1 line   Andrew Thomas
• [asterisk-users] cmd MySQL   Andrew Thomas
• [asterisk-users] calls are not going thru e1 line   Andrew Thomas
• [asterisk-users] CDR with unix time.   Robert Thomas
• [asterisk-users] fail-over server   Jonathan Thurman
• [asterisk-users] fail-over server   Jonathan Thurman
• [asterisk-users] Asterisk 1.8.3   Jonathan Thurman
• [asterisk-users] Cisco 7960 & asterisk 1.8.22 ringlist.dat error   Jonathan Thurman
• [asterisk-users] Cisco 7960 & asterisk 1.8.22 ringlist.dat error   Jonathan Thurman
• [asterisk-users] Cisco 7960 & asterisk 1.8.22 ringlist.dat error   Jonathan Thurman
• [asterisk-users] Realtime and Local Channel Crash Problem 1.8.3-rc2   Jonathan Thurman
• [asterisk-users] Manual Call Transfer // Perl // Asterisk::AGI // MySQL   Ted Tiberio
• [asterisk-users] Manual Call Transfer (Perl, Asterisk::AGI, MySQL)   Ted Tiberio
• [asterisk-users] queue called by agi doesn't re-enter the script   Ted Tiberio
• [asterisk-users] Fax Woes   Steve Totaro
• [asterisk-users] how to get Current Calls details   Daniel Tryba
• [asterisk-users] Email alerts for trunks (peers)   Daniel Tryba
• [asterisk-users] Dial command   Daniel Tryba
• [asterisk-users] Carrying context from one server to another?   Daniel Tryba
• [asterisk-users] Two Asterisk machines for redundancy   Daniel Tryba
• [asterisk-users] RTP keepalive doesn't work   Ryan Tucker
• [asterisk-users] Email alerts for trunks (peers)   Ryan Tucker
• [asterisk-users] Any voice changer applications for Asterisk?   Steve Underwood
• [asterisk-users] Any voice changer applications for Asterisk?   Steve Underwood
• [asterisk-users] Using voice modem as poor man's FXO in Asterisk 1.8   Steve Underwood
• [asterisk-users] Polycom Blf / Directed Pickup   Gord Urquhart
• [asterisk-users] exceeds the maximum size of ast_fdset error on Asterisk-1.8.0   RAJNIKANT VANZA
• [asterisk-users] remote bridging   Ondrej Valousek
• [asterisk-users] Codec negotiation   Ondrej Valousek
• [asterisk-users] SIP registration   Vieri
• [asterisk-users] fail-over server   Vieri
• [asterisk-users] fail-over server   Vieri
• [asterisk-users] fail-over server   Vieri
• [asterisk-users] pipe audio stream to external application   Vieri
• [asterisk-users] Polycom IP335   Ryan Wagoner
• [asterisk-users] Polycom IP335   Ryan Wagoner
• [asterisk-users] Polycom IP335   Ryan Wagoner
• [asterisk-users] Polycom IP335   Ryan Wagoner
• [asterisk-users] Polycom IP335   Ryan Wagoner
• [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?   Ryan Wagoner
• [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?   Ryan Wagoner
• [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?   Ryan Wagoner
• [asterisk-users] Paging with Polycom 3.3.x   Ryan Wagoner
• [asterisk-users] Need to buy the Digium card, to confirm   Ryan Wagoner
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   Watkins, Bradley
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   Watkins, Bradley
• [asterisk-users] function Echo() doesn't work   Karsten Wemheuer
• [asterisk-users] SIP MESSAGE outside calls - state of the art?   Roger Burton West
• [asterisk-users] meetme conference & playback of random sound file   Roger Burton West
• [asterisk-users] Using files .call or AMI   Roger Burton West
• [asterisk-users] Using files .call or AMI   Roger Burton West
• [asterisk-users] Using files .call or AMI   Roger Burton West
• [asterisk-users] On-Hold Music   Roger Burton West
• [asterisk-users] On-Hold Music   Roger Burton West
• [asterisk-users] On-Hold Music   Roger Burton West
• [asterisk-users] Unknown calls   Roger Burton West
• [asterisk-users] Carrying context from one server to another?   Roger Burton West
• [asterisk-users] Detect DTMF tone during call?   Roger Burton West
• [asterisk-users] How to disable srtp in asterisk 1.8.2.3?   Joe Williams
• [asterisk-users] Fax for Asterisk SIP-TDM   Mark Willis
• [asterisk-users] [modules.conf] Modules still loaded after "noload"   Mark Willis
• [asterisk-users] [modules.conf] Modules still loaded after "noload"   Mark Willis
• [asterisk-users] secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1   Terry Wilson
• [asterisk-users] AMI FullyBooted issue   Terry Wilson
• [asterisk-users] AMI FullyBooted issue   Terry Wilson
• [asterisk-users] Asterisk/Skype   Terry Wilson
• [asterisk-users] Asterisk/Skype   Terry Wilson
• [asterisk-users] SRTP Error Message   Terry Wilson
• [asterisk-users] 1.8.2.4: SIP dialogs not killed?   Terry Wilson
• [asterisk-users] 1.8.2.4: SIP dialogs not killed?   Terry Wilson
• [asterisk-users] Email alerts for trunks (peers)   Hans Witvliet
• [asterisk-users] uptime   Hans Witvliet
• [asterisk-users] uptime   Hans Witvliet
• [asterisk-users] Hide the plain text password   Hans Witvliet
• [asterisk-users] Hide the plain text password (suggestion)   Hans Witvliet
• [asterisk-users] Adjusting Rx and Tx gains   Hans Witvliet
• [asterisk-users] SIP Hardphone that works well with asterisk   Hans Witvliet
• [asterisk-users] Problem installing FXS module in old digium 4 channel tdm card   Greg Woods
• [asterisk-users] standalone NOTIFY message handling for Asterisk   Feng Xu
• [asterisk-users] standalone NOTIFY message handling for Asterisk   Feng Xu
• [asterisk-users] Play one audio file to the called part before the Dial() command‏   Songtao Yu
• [asterisk-users] further action after caller in a queue hangs up   Richard Zheng
• [asterisk-users] further action after caller in a queue hangs up   Richard Zheng
• [asterisk-users] Asterisk with TE 121 DADHI incoming calls fail   Jan Zieher
• [asterisk-users] Asterisk 1.8.3 BLF stopped working   Bryant Zimmerman
• [asterisk-users] Voicemail email attachment as MP3, with tags containing sender name, number, message number   Bryant Zimmerman
• [asterisk-users] Recieve_Fax caused crash 1.8.2.3   Bryant Zimmerman
• [asterisk-users] Voicemail email attachment as MP3, with tags containing sender name, number, message number   --[ UxBoD ]--
• [asterisk-users] [OT] Yealink IP Phones   --[ UxBoD ]--
• [asterisk-users] SRTP Error Message   --[ UxBoD ]--
• [asterisk-users] CEL and PGSQL   --[ UxBoD ]--
• [asterisk-users] CEL and PGSQL   --[ UxBoD ]--
• [asterisk-users] On-Hold Music   ayodele abejide
• [asterisk-users] On-Hold Music   ayodele abejide
• [asterisk-users] Dial command   ayodele abejide
• [asterisk-users] Dial command   ayodele abejide
• [asterisk-users] Connect Asterisk to a cell phone   abhinav anand
• [asterisk-users] DTMF not detected, time out   asterisk asterisk
• [asterisk-users] DTMF not detected, time out   asterisk asterisk
• [asterisk-users] DTMF not detected, time out   asterisk asterisk
• [asterisk-users] Problem in dialing out   asterisk asterisk
• [asterisk-users] Using voice modem as poor man's FXO in Asterisk 1.8   asterisk asterisk
• [asterisk-users] Error loading module 'cdr_radius.so'   bakko
• [asterisk-users] Connect Asterisk to a cell phone   bakko
• [asterisk-users] T1 channel audio control   mark calcagno
• [asterisk-users] FAX on PRI to MFCR2   leonimar cape
• [asterisk-users] FAX on PRI to MFCR2   leonimar cape
• [asterisk-users] Meet me recording   covici at ccs.covici.com
• [asterisk-users] sip trunk balancing   marek cervenka
• [asterisk-users] issue with some numbers   salaheddine elharit
• [asterisk-users] issue with some numbers   salaheddine elharit
• [asterisk-users] issue with some numbers   salaheddine elharit
• [asterisk-users] calls between iax and sip   salaheddine elharit
• [asterisk-users] calls between iax and sip   salaheddine elharit
• [asterisk-users] Need to buy the Digium card, to confirm   bilal ghayyad
• [asterisk-users] Two Asterisk machines for redundancy   bilal ghayyad
• [asterisk-users] Two Asterisk machines for redundancy   bilal ghayyad
• [asterisk-users] queue called by agi doesn't re-enter the script   gincantalupo
• [asterisk-users] ashishchauhan07oct at gmail.com sent you a movie ticket redeemable at more than 200 nation wide theatre chains   ashishchauhan07oct at gmail.com
• [asterisk-users] FAX on PRI to MFCR2   bayardo.sanchez at gmail.com
• [asterisk-users] missing argument on AGI   cbulist at gmail.com
• [asterisk-users] 1.8.2.4: SIP dialogs not killed?   isrlgb at gmail.com
• [asterisk-users] How to Change The Caller Position in Queue   shayne.alone at gmail.com
• [asterisk-users] how to get Current Calls details   shayne.alone at gmail.com
• [asterisk-users] what are QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY veriables?   shayne.alone at gmail.com
• [asterisk-users] SIP Hardphone that works well with asterisk   shayne.alone at gmail.com
• [asterisk-users] Detect #,* DTMF in dialplan   shayne.alone at gmail.com
• [asterisk-users] SIP Hardphone that works well with asterisk   ast guy
• [asterisk-users] SIP Hardphone that works well with asterisk   ast guy
• [asterisk-users] SIP Hardphone that works well with asterisk   ast guy
• [asterisk-users] SIP Hardphone that works well with asterisk   ast guy
• [asterisk-users] SIP Hardphone that works well with asterisk   ast guy
• [asterisk-users] Cisco 7945G phone with asterisk   ast guy
• [asterisk-users] Cisco 7945G phone with asterisk   ast guy
• [asterisk-users] MEMBERINTERFACE and MEMBERNAME questions   magnus.b at inputinterior.se
• [asterisk-users] MEMBERINTERFACE and MEMBERNAME questions   magnus.b at inputinterior.se
• [asterisk-users] AddQueueMember and stateinterface question   magnus.b at inputinterior.se
• [asterisk-users] regarding sip.conf and extensions.conf   viswavardhanreddy karna
• [asterisk-users] Regarding asterisk   viswavardhanreddy karna
• [asterisk-users] Regarding bob-invite-alice xml scenario   viswavardhanreddy karna
• [asterisk-users] Regarding error in asterisk 1.6.2.16....   viswavardhanreddy karna
• [asterisk-users] Regarding Asterisk   viswavardhanreddy karna
• [asterisk-users] forward calls by the ports   mehran khajavi
• [asterisk-users] REFER and dialplan broken (as documented in chan_sip.c on line 11951)   vip killa
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   vip killa
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   vip killa
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   vip killa
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   vip killa
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   vip killa
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   vip killa
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   vip killa
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   vip killa
• [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)   vip killa
• [asterisk-users] Connect Asterisk to a cell phone   logan
• [asterisk-users] Connect Asterisk to a cell phone   logan
• [asterisk-users] Asterisk on a USB with persistence   logan
• [asterisk-users] Looking for actual user opinions on Telephony card   john millican
• [asterisk-users] Outgoing FXO calls have no audio with callprogress=no   ftarz at mindspring.com
• [asterisk-users] Asterisk CallCompletion dialplan   satish patel
• [asterisk-users] Asterisk 1.8.3   satish patel
• [asterisk-users] Asterisk 1.8.3   satish patel
• [asterisk-users] Asterisk 1.8.3   satish patel
• [asterisk-users] Asterisk compile option DAHDI SPANS   satish patel
• [asterisk-users] Asterisk compile option DAHDI SPANS   satish patel
• [asterisk-users] asterisk 1.8.2 freez   satish patel
• [asterisk-users] pstack debug asterisk   satish patel
• [asterisk-users] Meet me recording   satish patel
• [asterisk-users] Meet me recording   satish patel
• [asterisk-users] Asterisk 1.2 zap hangup issue   satish patel
• [asterisk-users] asterisk security....again   satish patel
• [asterisk-users] Friday 18 Feb at 12 Noon EST: SylkServer and Blink   randulo
• [asterisk-users] SIP Hardphone that works well with asterisk   randulo
• [asterisk-users] AstriEurope coference   randulo
• [asterisk-users] AstriEurope coference   randulo
• [asterisk-users] Error loading module 'cdr_radius.so'   Safarifone Noc Technical Support s
• [asterisk-users] T.38 negotiation error   Marcello Colucci (SIRIO Informatica s.a.s.)
• [asterisk-users] R: T.38 negotiation error   Marcello Colucci (SIRIO Informatica s.a.s.)
• [asterisk-users] Outgoing agent´s calls   equis software
• [asterisk-users] Outgoing agent´s calls   equis software
• [asterisk-users] Outgoing agent´s calls   equis software
• [asterisk-users] standalone NOTIFY message handling for Asterisk   faisal at vopium.com
• [asterisk-users] Can a duration limit be specified in spool call file?   faisal at vopium.com
• [asterisk-users] Codec negotiation   faisal at vopium.com
• [asterisk-users] Call Recording audio file quality query   faisal at vopium.com
• [asterisk-users] Call Recording audio file quality query   faisal at vopium.com
• [asterisk-users] Call files error   faisal at vopium.com
• [asterisk-users] ${HANGUPCAUSE} in CDR   faisal at vopium.com
• [asterisk-users] Call files error   faisal at vopium.com
• [asterisk-users] Asterisk 1.8.3   Захаров Антон

Last message date: Mon Feb 28 23:20:03 CST 2011
Archived on: Mon Feb 28 23:22:09 CST 2011

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