July 2012 Archives by author

• Messages sorted by: [ thread ] [ subject ] [ date ]
• More info on this list...

Starting: Sun Jul 1 02:34:56 CDT 2012
Ending: Tue Jul 31 21:35:23 CDT 2012
Messages: 577

• [asterisk-users] How to play DTMF digits without blocking   ' Barros '
• [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used   Raj Mathur ( राज माथुर )
• [asterisk-users] Finding the position of a character in a string   Raj Mathur ( राज माथुर )
• [asterisk-users] CAS T1 - No Ringback   Raj Mathur ( राज माथुर )
• [asterisk-users] CAS T1 - No Ringback   Raj Mathur ( राज माथुर )
• [asterisk-users] best PRI gateway?   Raj Mathur ( राज माथुर )
• [asterisk-users] So long, and thanks for all the fish!   Raj Mathur ( राज माथुर )
• [asterisk-users] Reliable SIP Trunk Provider   Ignacio Ortega A.
• [asterisk-users] How to work around asterisk ss7   Ashish Agarwal
• [asterisk-users] How to work around asterisk ss7   Ashish Agarwal
• [asterisk-users] How to work around asterisk ss7   Ashish Agarwal
• [asterisk-users] touch command not behaving for future calls in asterisk 1.4.41   Gohar Ahmed
• [asterisk-users] weird dect beheaviour multiple handsets   Roland o/d Akker
• [asterisk-users] Play announcement during Dial just once   Roland o/d Akker
• [asterisk-users] Maximum concurrent calls using call files   Stephen J Alexander
• [asterisk-users] freepbx asterisk   Stephen J Alexander
• [asterisk-users] Call recording and hosted PBX platform using Asterisk   Carlos Alvarez
• [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash   Carlos Alvarez
• [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash   Carlos Alvarez
• [asterisk-users] Multi-Tenant PBX with Asterisk   Carlos Alvarez
• [asterisk-users] Multi-Tenant PBX with Asterisk   Carlos Alvarez
• [asterisk-users] Multi-Tenant PBX with Asterisk   Carlos Alvarez
• [asterisk-users] Multi-Tenant PBX with Asterisk   Carlos Alvarez
• [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used   Benny Amorsen
• [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used   Benny Amorsen
• [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used   Benny Amorsen
• [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used   Benny Amorsen
• [asterisk-users] Elastix 2.3.0.1   Satria Anamarta
• [asterisk-users] Trixbox or FreePBX?   Satria Anamarta
• [asterisk-users] Trixbox or FreePBX?   Satria Anamarta
• [asterisk-users] Route incoming calls   Satria Anamarta
• [asterisk-users] seems like call is picked and returned to me   Andres
• [asterisk-users] Asterisk with OpenBTS and mobile phone   Ellen Apolinar
• [asterisk-users] Asterisk with OpenBTS and mobile phone   Ellen Apolinar
• [asterisk-users] Asterisk with OpenBTS and mobile phone   Ellen Apolinar
• [asterisk-users] Video call using Asterisk   Julio Araujo
• [asterisk-users] Video call using Asterisk   Julio Araujo
• [asterisk-users] Video call using Asterisk   Julio Araujo
• [asterisk-users] Static noise on bridged calls to PSTN, although the trunk line is clean on its own   Sebastian Arcus
• [asterisk-users] any working calling card solution "open source"   Goke M Aruna
• [asterisk-users] any working calling card solution "open source"   Goke M Aruna
• [asterisk-users] Speech recognition in asterisk using google voice API   Bruce B
• [asterisk-users] asterisk crash   Brandon B.
• [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash   Chris Bagnall
• [asterisk-users] click to call   Chris Bagnall
• [asterisk-users] What TTS to use?   Chris Bagnall
• [asterisk-users] What TTS to use?   Chris Bagnall
• [asterisk-users] Digium IP Phone D40 quality, very bad   Chris Bagnall
• [asterisk-users] Less good call quality using Asterisk   Bakko
• [asterisk-users] How to send a SIP MESSAGE outside a call   Bakko
• [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?   Alex Balashov
• [asterisk-users] Regrading Speech Recognition.   Satish Barot
• [asterisk-users] Asterisk   Satish Barot
• [asterisk-users] How to set SIP to auto answer in the dial plan .   Jared Baxley
• [asterisk-users] Asterisk on Dynamic IP to a SIP extension   Jared Baxley
• [asterisk-users] Binary packages for Ubuntu Precise   Paul Belanger
• [asterisk-users] Recommended VOIP Monitoring Tools   Paul Belanger
• [asterisk-users] Asterisk on Dynamic IP to a SIP extension   Paul Belanger
• [asterisk-users] Digium IP Phone D40 quality, very bad   Paul Belanger
• [asterisk-users] DAHDI DTMF problem?   Ron Bergin
• [asterisk-users] DAHDI DTMF problem?   Ron Bergin
• [asterisk-users] How to set SIP to auto answer in the dial plan .   Ron Bergin
• [asterisk-users] echo canceler query   Agustina Berretta
• [asterisk-users] HP DL360 G5 better than HP DL360 G7 ?   Tom Browning
• [asterisk-users] HP DL360 G5 better than HP DL360 G7 ?   Tom Browning
• [asterisk-users] Queue Member login from IAX trunk   Jakob-Matthias Böttger
• [asterisk-users] Queue Member login from IAX trunk   Jakob-Matthias Böttger
• [asterisk-users] Asterisk 1.8.13 PlayTones App   Jakob-Matthias Böttger
• [asterisk-users] AGI not generating sip 180/183 status   Marek Cervenka
• [asterisk-users] Video conferencing?   Carlos Chavez
• [asterisk-users] file and on SayNumber() app   Tzafrir Cohen
• [asterisk-users] file and on SayNumber() app   Tzafrir Cohen
• [asterisk-users] problem with 8 port card   Tzafrir Cohen
• [asterisk-users] libpri error   Tzafrir Cohen
• [asterisk-users] Dahdi Dropping Calls   Andrew Colin
• [asterisk-users] Centos 6 mISDN   Andrew Colin
• [asterisk-users] basic sip quesiton   Andrew Colin
• [asterisk-users] Centos 6 mISDN   Andrew Colin
• [asterisk-users] NO AUDIO   Thiago Coutinho
• [asterisk-users] Changing auto mixmonitor output file name   Thiago Coutinho
• [asterisk-users] Agent receives call while making calls   Thiago Coutinho
• [asterisk-users] So long, and thanks for all the fish!   Thiago Coutinho
• [asterisk-users] SendFAX timestamp   David Cunningham
• [asterisk-users] Video conferencing?   Ken D'Ambrosio
• [asterisk-users] Video conferencing?   Ken D'Ambrosio
• [asterisk-users] Video conferencing (and SMTP server hiccups)?   Ken D'Ambrosio
• [asterisk-users] port 5060 is blocked by ISP   Leandro Dardini
• [asterisk-users] trouble with asterisk behind router   Leandro Dardini
• [asterisk-users] trouble with asterisk behind router   Leandro Dardini
• [asterisk-users] PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk   Leandro Dardini
• [asterisk-users] PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk   Leandro Dardini
• [asterisk-users] Call ID of the second call leg   Leandro Dardini
• [asterisk-users] Call ID of the second call leg   Leandro Dardini
• [asterisk-users] Call ID of the second call leg   Leandro Dardini
• [asterisk-users] Multiple DID for SIP Trunk   Leandro Dardini
• [asterisk-users] Multi-Tenant PBX with Asterisk   Leandro Dardini
• [asterisk-users] Multi-Tenant PBX with Asterisk   Leandro Dardini
• [asterisk-users] Multi-Tenant PBX with Asterisk   Leandro Dardini
• [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used   Alec Davis
• [asterisk-users] SIP Inband DTMF problem   Kalin Dikov
• [asterisk-users] Voicemail Emails   Danilo Dionisi
• [asterisk-users] Asterisk on Dynamic IP to a SIP extension   Doug
• [asterisk-users] As Kevin Fleming says "So long, and thanks for all the fish!", we say thank you - and look to the future   David Duffett
• [asterisk-users] Audiocodes 310HD - on Asterisk Server   Paul Edgar
• [asterisk-users] Agent receives call while making calls   Eduardo
• [asterisk-users] call file and NFS server   Steve Edwards
• [asterisk-users] call file and NFS server   Steve Edwards
• [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration   Steve Edwards
• [asterisk-users] How to set SIP to auto answer in the dial plan .   Steve Edwards
• [asterisk-users] IAX trunking stopped working   Noah Engelberth
• [asterisk-users] IAX trunking stopped working   Noah Engelberth
• [asterisk-users] res_odbc crashing asterisk after freetds dsn reconnects   Noah Engelberth
• [asterisk-users] OT - Multi Function Printer with one touch scanning/emailing   C F
• [asterisk-users] OT - Multi Function Printer with one touch scanning/emailing   C F
• [asterisk-users] OT - Multi Function Printer with one touch scanning/emailing   C F
• [asterisk-users] Outbound Asterisk calls default directmedia specifications   Kevin P. Fleming
• [asterisk-users] Asterisk trying to call a queue with no members   Kevin P. Fleming
• [asterisk-users] Asterisk trying to call a queue with no members   Kevin P. Fleming
• [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash   Kevin P. Fleming
• [asterisk-users] sip.conf and binaddr issue   Kevin P. Fleming
• [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash   Kevin P. Fleming
• [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used   Kevin P. Fleming
• [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used   Kevin P. Fleming
• [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used   Kevin P. Fleming
• [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used   Kevin P. Fleming
• [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used   Kevin P. Fleming
• [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway   Kevin P. Fleming
• [asterisk-users] Asterisk 1.8.13 / res_fax / res_fax_digium   Kevin P. Fleming
• [asterisk-users] Asterisk 1.8.13 / res_fax / res_fax_digium   Kevin P. Fleming
• [asterisk-users] Channel is rsrvd and does not turn off   Kevin P. Fleming
• [asterisk-users] T.38 (PRI) Fax Debugging   Kevin P. Fleming
• [asterisk-users] Voicemail Emails   Kevin P. Fleming
• [asterisk-users] Voicemail Emails   Kevin P. Fleming
• [asterisk-users] T.38 Gateway   Kevin P. Fleming
• [asterisk-users] Less good call quality using Asterisk   Kevin P. Fleming
• [asterisk-users] Less good call quality using Asterisk   Kevin P. Fleming
• [asterisk-users] Polycom Presence with Asterisk 1.8.12.0   Kevin P. Fleming
• [asterisk-users] still got ReceiveFax() problem, how to properly setup asterisk fax?   Kevin P. Fleming
• [asterisk-users] still got ReceiveFax() problem, how to properly setup asterisk fax?   Kevin P. Fleming
• [asterisk-users] best PRI gateway?   Kevin P. Fleming
• [asterisk-users] So long, and thanks for all the fish!   Kevin P. Fleming
• [asterisk-users] NO AUDIO   Daviramos Roussenq Fortunato
• [asterisk-users] Polycom Presence with Asterisk 1.8.12.0   Dave Fullerton
• [asterisk-users] Telecom HU cannot callforward to external number   Christian Gansberger
• [asterisk-users] Telecom HU cannot callforward to external number   Christian Gansberger
• [asterisk-users] connections to manager   Jerry Geis
• [asterisk-users] connections to manager   Jerry Geis
• [asterisk-users] Asterisk 1.8.15.0 Now Available   Eric Germann
• [asterisk-users] SIP/GSM-gateway recommendation?   Stefan Gofferje
• [asterisk-users] T.38 Gateway   Paul Goldbaum
• [asterisk-users] Please dont tell me this is impossible   Thorsten Göllner
• [asterisk-users] Please dont tell me this is impossible   Thorsten Göllner
• [asterisk-users] Please dont tell me this is impossible   Thorsten Göllner
• [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used   Freddi Hansen
• [asterisk-users] asterisk-users list testing - hopefull2 at hotmail.com   Hopefull2
• [asterisk-users] Voicemail Emails   Josh Hopkins
• [asterisk-users] Voicemail Emails   Josh Hopkins
• [asterisk-users] T.30 Fax session error: Received bad response to DCS or training   Lee Howard
• [asterisk-users] Asterisk 1.8.12 and Fax?   Lee Howard
• [asterisk-users] Allworx 9212   Hutter, Mark
• [asterisk-users] IAX2 Registered OK without IP   Alejandro Imass
• [asterisk-users] IAX2 Registered OK without IP   Alejandro Imass
• [asterisk-users] Asterisk with OpenBTS and mobile phone   Ioan Indreias
• [asterisk-users] WOW Offer from DIDX for this week!!   Muneeb Iqbal
• [asterisk-users] WOW Offer from DIDX for this week!!   Muneeb Iqbal
• [asterisk-users] WOW Offer from DIDX for this week!!   Muneeb Iqbal
• [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?   Ira
• [asterisk-users] Getting error while dialing outside (PRI Connection card TE110P ASTERISK1.8 AND FREEPBX 2.10   Darin Iv
• [asterisk-users] MixMonitor creating file on non-bridged calls with option b   Thorben Jensen
• [asterisk-users] Intro to DECT vs IP   Andrew Joakimsen
• [asterisk-users] Forcing SIP trunk matching order?   Olle E. Johansson
• [asterisk-users] port 5060 is blocked by ISP   Olle E. Johansson
• [asterisk-users] Timer1 RFC and SIP.CONF   Olle E. Johansson
• [asterisk-users] Timer1 RFC and SIP.CONF   Olle E. Johansson
• [asterisk-users] sip.conf and binaddr issue   Olle E. Johansson
• [asterisk-users] Rookie / sip and extensions   Olle E. Johansson
• [asterisk-users] seems like call is picked and returned to me   Olle E. Johansson
• [asterisk-users] sip.conf and binaddr issue   Olle E. Johansson
• [asterisk-users] Forcing SIP trunk matching order?   Olle E. Johansson
• [asterisk-users] IAX trunking stopped working   Matthew Jordan
• [asterisk-users] FreePBX: How to hangup if the caller did not press # after the voicemail message   Matthew Jordan
• [asterisk-users] SRTP Encryption Per Device   Matthew Jordan
• [asterisk-users] 10.6.0-rc2: tmp full of core.PBX   Matthew Jordan
• [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration   Matthew Jordan
• [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway   Matthew Jordan
• [asterisk-users] Inconsistency in CDR between NO ANSWER and BUSY calls   Matthew Jordan
• [asterisk-users] res_odbc crashing asterisk after freetds dsn reconnects   Matthew Jordan
• [asterisk-users] Video conferencing?   Matthew Jordan
• [asterisk-users] No audio playing back voicemail from odbc   Matthew Jordan
• [asterisk-users] Asterisk + Google Voice   Matthew Jordan
• [asterisk-users] How to send a SIP MESSAGE outside a call   Matthew Jordan
• [asterisk-users] How to send a SIP MESSAGE outside a call   Matthew Jordan
• [asterisk-users] No audio playing back voicemail from odbc   Matthew Jordan
• [asterisk-users] How to send a SIP MESSAGE outside a call   Matthew Jordan
• [asterisk-users] No audio playing back voicemail from odbc   Matthew Jordan
• [asterisk-users] Call recording and transfer issue (asterisk 1.8)   Matthew Jordan
• [asterisk-users] Video conferencing?   Matthew Jordan
• [asterisk-users] call file and NFS server   Arstan Jusupov
• [asterisk-users] Route incoming calls   Arstan Jusupov
• [asterisk-users] Asterisk Capacity   Kannan
• [asterisk-users] PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk   Kannan
• [asterisk-users] PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk   Kannan
• [asterisk-users] PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk   Kannan
• [asterisk-users] PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk   Kannan
• [asterisk-users] PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk   Kannan
• [asterisk-users] Multiple DID for SIP Trunk   Kannan
• [asterisk-users] Multi-Tenant PBX with Asterisk   Kannan
• [asterisk-users] Multi-Tenant PBX with Asterisk   Kannan
• [asterisk-users] Multi-Tenant PBX with Asterisk   Kannan
• [asterisk-users] Realtime Queue and Queue_members   Jonas Kellens
• [asterisk-users] port 5060 is blocked by ISP   Thomas Kenyon
• [asterisk-users] asterisk 1.8 on Solaris/sparc   Jeremy Kister
• [asterisk-users] asterisk 1.8 on Solaris/sparc   Jeremy Kister
• [asterisk-users] asterisk 1.8 on Solaris/sparc   Jeremy Kister
• [asterisk-users] QoS : tos and cos settings   Klaverstyn, David C
• [asterisk-users] Does the Eicon/Dialogic 4BRI-8M 800-665 support the NT modus?   Michelle Konzack
• [asterisk-users] Does the Eicon/Dialogic 4BRI-8M 800-665 support the NT modus?   Michelle Konzack
• [asterisk-users] Can not get my Eicon Diva running with Asterisk...   Michelle Konzack
• [asterisk-users] SIP/GSM-gateway recommendation?   Michelle Konzack
• [asterisk-users] any working calling card solution "open source"   Stelios Koroneos
• [asterisk-users] PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk   Stelios Koroneos
• [asterisk-users] libpri error   Kamlesh Kumar
• [asterisk-users] libpri error   Kamlesh Kumar
• [asterisk-users] libpri error   Kamlesh Kumar
• [asterisk-users] libpri error   Kamlesh Kumar
• [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used   Jeff LaCoursiere
• [asterisk-users] How to Auto Answer a sip phone   Bharat Lalcheta
• [asterisk-users] How to Auto Answer a sip phone   Bharat Lalcheta
• [asterisk-users] How to work around asterisk ss7   Bharat Lalcheta
• [asterisk-users] MWI not working - Asterisk 1.8.9.2   Bharat Lalcheta
• [asterisk-users] Forcing SIP trunk matching order?   James Lamanna
• [asterisk-users] Channel is rsrvd and does not turn off   Rodrigo Lang
• [asterisk-users] Channel is rsrvd and does not turn off   Rodrigo Lang
• [asterisk-users] Channel is rsrvd and does not turn off   Rodrigo Lang
• [asterisk-users] Elastix 2.3.0.1   Alex Villací­s Lasso
• [asterisk-users] Regrading Speech Recognition.   Mitul Limbani
• [asterisk-users] Telecom HU cannot callforward to external number   Mitul Limbani
• [asterisk-users] How to work around asterisk ss7   Mitul Limbani
• [asterisk-users] How to work around asterisk ss7   Mitul Limbani
• [asterisk-users] PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk   Mitul Limbani
• [asterisk-users] PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk   Mitul Limbani
• [asterisk-users] DAHDI problems   Mitul Limbani
• [asterisk-users] DAHDI problems   Mitul Limbani
• [asterisk-users] passing arguments to macros from originate command   Mitul Limbani
• [asterisk-users] Asterisk   Mitul Limbani
• [asterisk-users] Dahdi+Redfone+Channel Bank+E&M   Mitul Limbani
• [asterisk-users] Dahdi+Redfone+Channel Bank+E&M   Mitul Limbani
• [asterisk-users] CAS T1 - No Ringback   Mitul Limbani
• [asterisk-users] Multiple DID for SIP Trunk   Mitul Limbani
• [asterisk-users] best PRI gateway?   Mitul Limbani
• [asterisk-users] Rookie / sip and extensions   Mikhail Lischuk
• [asterisk-users] Meeting SIP and RTSP   Mikhail Lischuk
• [asterisk-users] Centos 6 mISDN   Patrick Lists
• [asterisk-users] AMR - Segmentation Fault   Patrick Lists
• [asterisk-users] AMR - Segmentation Fault   Patrick Lists
• [asterisk-users] Centos 6 mISDN   Patrick Lists
• [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?   Patrick Lists
• [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?   Patrick Lists
• [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?   Patrick Lists
• [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?   Patrick Lists
• [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?   Patrick Lists
• [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?   Patrick Lists
• [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash   Patrick Lists
• [asterisk-users] freepbx asterisk   Patrick Lists
• [asterisk-users] What TTS to use?   Patrick Lists
• [asterisk-users] after upgrade from 1.4.26.2 to 1.8.11.0 my provider gets rport instead of port   Shitian Long
• [asterisk-users] OT - Integration with building intercom systems   Shitian Long
• [asterisk-users] sip and extensions   Shitian Long
• [asterisk-users] How I could Insert a record into the table with REALTIME   Shitian Long
• [asterisk-users] How to set SIP to auto answer in the dial plan .   Doug Lytle
• [asterisk-users] Issue with a ticket system subscribed to asterisk-users   Doug Lytle
• [asterisk-users] Getting error while dialing outside (PRI Connection card TE110P ASTERISK1.8 AND FREEPBX 2.10   Doug Lytle
• [asterisk-users] Remote party ID - sort of working...   Doug Lytle
• [asterisk-users] Less good call quality using Asterisk   Doug Lytle
• [asterisk-users] Confbridge examples for Asterisk 10?   Doug Lytle
• [asterisk-users] BT Fibre and 2701HGV   Ishfaq Malik
• [asterisk-users] Inconsistency in CDR between NO ANSWER and BUSY calls   Ishfaq Malik
• [asterisk-users] Inconsistency in CDR between NO ANSWER and BUSY calls   Ishfaq Malik
• [asterisk-users] Inconsistency in CDR between NO ANSWER and BUSY calls   Ishfaq Malik
• [asterisk-users] Call recording and transfer issue (asterisk 1.8)   Ishfaq Malik
• [asterisk-users] Finding the position of a character in a string   Ishfaq Malik
• [asterisk-users] Finding the position of a character in a string   Ishfaq Malik
• [asterisk-users] What TTS to use?   Ishfaq Malik
• [asterisk-users] What TTS to use?   Ishfaq Malik
• [asterisk-users] Asterisk Realtime issue after registering withx-lite   Ishfaq Malik
• [asterisk-users] Multi-Tenant PBX with Asterisk   Ishfaq Malik
• [asterisk-users] Call recording and transfer issue (asterisk 1.8)   Ishfaq Malik
• [asterisk-users] Distributed Device Status with XMPP PubSub   Andrew McRory
• [asterisk-users] Video conferencing?   Dmitry Melekhov
• [asterisk-users] Video conferencing?   Dmitry Melekhov
• [asterisk-users] Dahdi+Redfone+Channel Bank+E&M   Jorge Mendoza
• [asterisk-users] Dahdi+Redfone+Channel Bank+E&M   Jorge Mendoza
• [asterisk-users] Dahdi+Redfone+Channel Bank+E&M   Jorge Mendoza
• [asterisk-users] DAHDI problems   Russ Meyerriecks
• [asterisk-users] DAHDI problems   Russ Meyerriecks
• [asterisk-users] problem with 8 port card   Russ Meyerriecks
• [asterisk-users] - SIP retransmission problem   Paolo De Michele
• [asterisk-users] click to call   Mike
• [asterisk-users] click to call   Mike
• [asterisk-users] Recommended VOIP Monitoring Tools   Mike
• [asterisk-users] QoS : tos and cos settings   Mike
• [asterisk-users] Remote party ID - sort of working...   Mike
• [asterisk-users] Remote party ID - sort of working...   Mike
• [asterisk-users] Less good call quality using Asterisk   Mike
• [asterisk-users] best PRI gateway?   Mike
• [asterisk-users] Asterisk trying to call a queue with no members   Antonio Modesto
• [asterisk-users] Asterisk trying to call a queue with no members   Antonio Modesto
• [asterisk-users] Asterisk trying to call a queue with no members   Antonio Modesto
• [asterisk-users] Queue timeoutpriority=app doesn't working as explained in conf.sample   Antonio Modesto
• [asterisk-users] Changing auto mixmonitor output file name   Antonio Modesto
• [asterisk-users] T.30 Fax session error: Received bad response to DCS or training   Petros Moisiadis
• [asterisk-users] How to set SIP to auto answer in the dial plan .   Larry Moore
• [asterisk-users] Remote party ID - sort of working...   Richard Mudgett
• [asterisk-users] Remote party ID - sort of working...   Richard Mudgett
• [asterisk-users] callback on busy   Richard Mudgett
• [asterisk-users] Timer1 RFC and SIP.CONF   Elliot Murdock
• [asterisk-users] Timer1 RFC and SIP.CONF   Elliot Murdock
• [asterisk-users] SRTP Encryption Per Device   Elliot Murdock
• [asterisk-users] Recommended VOIP Monitoring Tools   Elliot Murdock
• [asterisk-users] sip and extensions   Tim Nelson
• [asterisk-users] Can I install Asterisk normally and then installing the GUI "asterisk now"   Tim Nelson
• [asterisk-users] DAHDI DTMF problem?   Tim Nelson
• [asterisk-users] DAHDI DTMF problem?   Tim Nelson
• [asterisk-users] Can I install Asterisk normally and then installing the GUI   Tim Nelson
• [asterisk-users] Trixbox or FreePBX?   Tim Nelson
• [asterisk-users] Trixbox or FreePBX?   Tim Nelson
• [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash   Tim Nelson
• [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash   Tim Nelson
• [asterisk-users] Polycom Presence with Asterisk 1.8.12.0   Tim Nelson
• [asterisk-users] Polycom Presence with Asterisk 1.8.12.0   Tim Nelson
• [asterisk-users] CAS T1 - No Ringback   Tim Nelson
• [asterisk-users] CAS T1 - No Ringback   Tim Nelson
• [asterisk-users] Issue with a ticket system subscribed to asterisk-users   Rusty Newton
• [asterisk-users] Issue with a ticket system subscribed to asterisk-users   Rusty Newton
• [asterisk-users] WOW Offer from DIDX for this week!!   Rusty Newton
• [asterisk-users] WOW Offer from DIDX for this week!!   Rusty Newton
• [asterisk-users] T.38 Gateway   Rusty Newton
• [asterisk-users] asterisk crash   Rusty Newton
• [asterisk-users] Digium IP Phone D40 quality, very bad   Rusty Newton
• [asterisk-users] How to play different different hold music.   Danny Nicholas
• [asterisk-users] SRTP Encryption Per Device   Danny Nicholas
• [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?   Danny Nicholas
• [asterisk-users] click to call   Danny Nicholas
• [asterisk-users] Voicemail Emails   Danny Nicholas
• [asterisk-users] callback on busy   Danny Nicholas
• [asterisk-users] callback - disa   Danny Nicholas
• [asterisk-users] Polycom Presence with Asterisk 1.8.12.0   Danny Nicholas
• [asterisk-users] Asterisk on Dynamic IP to a SIP extension   John Novack
• [asterisk-users] OT - Integration with building intercom systems   Olivier
• [asterisk-users] OT - Multi Function Printer with one touch scanning/emailing   Olivier
• [asterisk-users] OT - Integration with building intercom systems   Olivier
• [asterisk-users] OT - Multi Function Printer with one touch scanning/emailing   Olivier
• [asterisk-users] OT - Multi Function Printer with one touch scanning/emailing   Olivier
• [asterisk-users] port 5060 is blocked by ISP   B. van Ouwerkerk
• [asterisk-users] file and on SayNumber() app   Yaroslav Panych
• [asterisk-users] basic sip quesiton   Thomas Perron
• [asterisk-users] sip and extensions   Thomas Perron
• [asterisk-users] sip and extensions   Thomas Perron
• [asterisk-users] Rookie / sip and extensions   Thomas Perron
• [asterisk-users] Rookie / sip and extensions   Thomas Perron
• [asterisk-users] trouble with asterisk behind router   Nikolay G. Petrov
• [asterisk-users] Recommended VOIP Monitoring Tools   Nikolay G. Petrov
• [asterisk-users] trouble with asterisk behind router   Nikolay G. Petrov
• [asterisk-users] trouble with asterisk behind router   Nikolay G. Petrov
• [asterisk-users] chan_sip sending from wrong source, address when multiple interfaces are used   Dave Platt
• [asterisk-users] Asterisk   Herve Prime
• [asterisk-users] asterisk-users list testing - msegovia.ins at gmail.com   Maria Segovia R.
• [asterisk-users] How to set SIP to auto answer in the dial plan .   Zohair Raza
• [asterisk-users] How to set SIP to auto answer in the dial plan .   Zohair Raza
• [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway   Alejandro Recarey
• [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway   Alejandro Recarey
• [asterisk-users] Recommended VOIP Monitoring Tools   Robert-IPhone
• [asterisk-users] Recommended VOIP Monitoring Tools   Robert-IPhone
• [asterisk-users] weird dect beheaviour multiple handsets   Carlos Rojas
• [asterisk-users] any working calling card solution "open source"   Carlos Rojas
• [asterisk-users] Video conferencing?   Jonathan Rose
• [asterisk-users] DAHDI DTMF problem?   Shaun Ruffell
• [asterisk-users] DAHDI DTMF problem?   Shaun Ruffell
• [asterisk-users] DAHDI DTMF problem?   Shaun Ruffell
• [asterisk-users] sip.conf and binaddr issue   Felix Salfelder
• [asterisk-users] sip.conf and bindaddr issue   Felix Salfelder
• [asterisk-users] port 5060 is blocked by ISP   SamyGo
• [asterisk-users] How to play different different hold music.   SamyGo
• [asterisk-users] How to play different different hold music.   SamyGo
• [asterisk-users] Outbound Asterisk calls default directmedia specifications   SamyGo
• [asterisk-users] How to play different different hold music.   SamyGo
• [asterisk-users] port 5060 is blocked by ISP   SamyGo
• [asterisk-users] Queue Member login from IAX trunk   SamyGo
• [asterisk-users] Queue Member login from IAX trunk   SamyGo
• [asterisk-users] sip set debug on always showing error   SamyGo
• [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration   SamyGo
• [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration   SamyGo
• [asterisk-users] touch command not behaving for future calls in asterisk 1.4.41   SamyGo
• [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration   SamyGo
• [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration   SamyGo
• [asterisk-users] PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk   SamyGo
• [asterisk-users] Asterisk + Google Voice   Rafael dos Santos Saraiva
• [asterisk-users] Asterisk + Google Voice   Rafael dos Santos Saraiva
• [asterisk-users] Does the Eicon/Dialogic 4BRI-8M 800-665 support the NT modus?   Armin Schindler
• [asterisk-users] Does the Eicon/Dialogic 4BRI-8M 800-665 support the NT modus?   Armin Schindler
• [asterisk-users] basic sip quesiton   Warren Selby
• [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration   Warren Selby
• [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration   Warren Selby
• [asterisk-users] Rookie / sip and extensions   Warren Selby
• [asterisk-users] channel not available and jump to next group channels   Warren Selby
• [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?   Warren Selby
• [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?   Warren Selby
• [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash   Warren Selby
• [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration   Warren Selby
• [asterisk-users] Voicemail Emails   Warren Selby
• [asterisk-users] seems like call is picked and returned to me   Sergio Serrano
• [asterisk-users] DAHDI DTMF problem?   Bill Dunn - VCI Internet Services
• [asterisk-users] DAHDI DTMF problem?   Bill Dunn - VCI Internet Services
• [asterisk-users] DAHDI DTMF problem?   Bill Dunn - VCI Internet Services
• [asterisk-users] DAHDI DTMF problem?   Bill Dunn - VCI Internet Services
• [asterisk-users] Call recording and hosted PBX platform using Asterisk   James Sharp
• [asterisk-users] How to Auto Answer a sip phone   James Sharp
• [asterisk-users] How to Auto Answer a sip phone   James Sharp
• [asterisk-users] 111 Useful and/or Funny New Prompts For Asterisk, Courtesy of Allison Smith   Steve Sokol
• [asterisk-users] AMR - Segmentation Fault   Chandrakant Solanki
• [asterisk-users] AMR - Segmentation Fault   Chandrakant Solanki
• [asterisk-users] call file and NFS server   Chandrakant Solanki
• [asterisk-users] call file and NFS server   Chandrakant Solanki
• [asterisk-users] AMR - Segmentation Fault   Chandrakant Solanki
• [asterisk-users] QUEUEMEMBER_STATUS incorrect?   Chet W. Stevens
• [asterisk-users] How to play different different hold music.   A J Stiles
• [asterisk-users] call file and NFS server   A J Stiles
• [asterisk-users] click to call   A J Stiles
• [asterisk-users] click to call   A J Stiles
• [asterisk-users] asterisk 1.8 on Solaris/sparc   A J Stiles
• [asterisk-users] Digium Phones: Heads Up   A J Stiles
• [asterisk-users] asterisk 1.8 on Solaris/sparc   A J Stiles
• [asterisk-users] (no subject)   A J Stiles
• [asterisk-users] libpri error   A J Stiles
• [asterisk-users] libpri error   A J Stiles
• [asterisk-users] libpri error   A J Stiles
• [asterisk-users] libpri error   A J Stiles
• [asterisk-users] Digium IP Phone D40 quality, very bad   A J Stiles
• [asterisk-users] still got ReceiveFax() problem, how to properly setup asterisk fax?   Roi Stork
• [asterisk-users] still got ReceiveFax() problem, how to properly setup asterisk fax?   Roi Stork
• [asterisk-users] No audio playing back voicemail from odbc   Support
• [asterisk-users] No audio playing back voicemail from odbc   Support
• [asterisk-users] OT - Integration with building intercom systems   Administrator TOOTAI
• [asterisk-users] Asterisk 1.8.11-cert4, 1.8.13.1, 10.5.2, 10.5.2-digiumphones Now Available (Security Release)   Asterisk Development Team
• [asterisk-users] Asterisk 1.8.14.0 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 10.6.0 Now Available   Asterisk Development Team
• [asterisk-users] Planned service outage for community services on July 12, 2012   Asterisk Development Team
• [asterisk-users] Asterisk 1.8.14.1 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 10.6.1 Now Available   Asterisk Development Team
• [asterisk-users] Unplanned service outage within next hour for Asterisk community services   Asterisk Development Team
• [asterisk-users] Asterisk 1.8.15.0 Now Available   Asterisk Development Team
• [asterisk-users] Asterisk 10.7.0 Now Available   Asterisk Development Team
• [asterisk-users] AST-2012-010: Possible resource leak on uncompleted re-invite transactions   Asterisk Security Team
• [asterisk-users] AST-2012-011: Remote crash vulnerability in voice mail application   Asterisk Security Team
• [asterisk-users] Video conferencing?   Steve Totaro
• [asterisk-users] OT - Multi Function Printer with one touch scanning/emailing   Rob Townley
• [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration   Duncan Turnbull
• [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used   Duncan Turnbull
• [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration   Duncan Turnbull
• [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration   Duncan Turnbull
• [asterisk-users] DAHDI problems   Duncan Turnbull
• [asterisk-users] IAX2 Registered OK without IP   Duncan Turnbull
• [asterisk-users] IAX2 Registered OK without IP   Duncan Turnbull
• [asterisk-users] callback on busy   Duncan Turnbull
• [asterisk-users] So long, and thanks for all the fish!   Duncan Turnbull
• [asterisk-users] SendFAX timestamp   Steve Underwood
• [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway   Steve Underwood
• [asterisk-users] AMR - Segmentation Fault   Giacomo Vacca
• [asterisk-users] How to send a SIP MESSAGE outside a call   Tiago Vasconcelos
• [asterisk-users] How to send a SIP MESSAGE outside a call   Tiago Vasconcelos
• [asterisk-users] How to send a SIP MESSAGE outside a call   Tiago Vasconcelos
• [asterisk-users] Less good call quality using Asterisk   Stefan at WPF
• [asterisk-users] Less good call quality using Asterisk   Stefan at WPF
• [asterisk-users] Less good call quality using Asterisk   Stefan at WPF
• [asterisk-users] FreePBX: How to hangup if the caller did not press # after the voicemail message   Eric Wieling
• [asterisk-users] channel not available and jump to next group channels   Eric Wieling
• [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash   Eric Wieling
• [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration   Eric Wieling
• [asterisk-users] Asterisk 1.8.13 / res_fax / res_fax_digium   Eric Wieling
• [asterisk-users] Asterisk 1.8.13 / res_fax / res_fax_digium   Eric Wieling
• [asterisk-users] Remote party ID - sort of working...   Eric Wieling
• [asterisk-users] Remote party ID - sort of working...   Eric Wieling
• [asterisk-users] T.38 (PRI) Fax Debugging   Eric Wieling
• [asterisk-users] T.38 (PRI) Fax Debugging   Eric Wieling
• [asterisk-users] still got ReceiveFax() problem, how to properly setup asterisk fax?   Eric Wieling
• [asterisk-users] best PRI gateway?   Eric Wieling
• [asterisk-users] just did sched_add waitid Warnings 1.8.14.1   Eric Wieling
• [asterisk-users] port 5060 is blocked by ISP   Hans Witvliet
• [asterisk-users] AMR - Segmentation Fault   Hans Witvliet
• [asterisk-users] AMR - Segmentation Fault   Hans Witvliet
• [asterisk-users] asterisk 1.8 on Solaris/sparc   Hans Witvliet
• [asterisk-users] HP DL360 G5 better than HP DL360 G7 ?   Hans Witvliet
• [asterisk-users] PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk   Hans Witvliet
• [asterisk-users] Speech recognition in asterisk using google voice API   Lefteris Zafiris
• [asterisk-users] Multi-Tenant PBX with Asterisk   Bryant Zimmerman
• [asterisk-users] (no subject)   aa aa
• [asterisk-users] (no subject)   aa aa
• [asterisk-users] touch command not behaving for future calls in asterisk 1.4.41   virendra bhati
• [asterisk-users] touch command not behaving for future calls in asterisk 1.4.41   virendra bhati
• [asterisk-users] What TTS to use?   virendra bhati
• [asterisk-users] Asterisk Realtime issue after registering with x-lite   virendra bhati
• [asterisk-users] Asterisk Realtime issue after registering withx-lite   virendra bhati
• [asterisk-users] Asterisk Realtime issue after registering withx-lite   virendra bhati
• [asterisk-users] 2GB Elastix memory limit   research at businesstz.com
• [asterisk-users] How to play different different hold music.   akhilesh chand
• [asterisk-users] How to play different different hold music.   akhilesh chand
• [asterisk-users] How to play different different hold music.   akhilesh chand
• [asterisk-users] How to play different different hold music.   akhilesh chand
• [asterisk-users] How to play different different hold music.   akhilesh chand
• [asterisk-users] Regrading Speech Recognition.   akhilesh chand
• [asterisk-users] Regrading Speech Recognition.   akhilesh chand
• [asterisk-users] (no subject)   akhilesh chand
• [asterisk-users] problem with 8 port card   akhilesh chand
• [asterisk-users] (no subject)   akhilesh chand
• [asterisk-users] Confbridge examples for Asterisk 10?   cjwstudios
• [asterisk-users] Confbridge examples for Asterisk 10?   cjwstudios
• [asterisk-users] 10.6.0-rc2: tmp full of core.PBX   sean darcy
• [asterisk-users] 10.6.0-rc2: tmp full of core.PBX   sean darcy
• [asterisk-users] Free PBX: hangup even if did not dial # in the voicemail   bilal ghayyad
• [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration   bilal ghayyad
• [asterisk-users] FreePBX: How to hangup if the caller did not press # after the voicemail message   bilal ghayyad
• [asterisk-users] Can I install Asterisk normally and then installing the GUI "asterisk now"   bilal ghayyad
• [asterisk-users] Can I install Asterisk normally and then installing the GUI   bilal ghayyad
• [asterisk-users] Trixbox or FreePBX?   bilal ghayyad
• [asterisk-users] FreePBX: How to hangup if the caller did not press # after the voicemail message   bilal ghayyad
• [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration   bilal ghayyad
• [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash   bilal ghayyad
• [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration   bilal ghayyad
• [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration   bilal ghayyad
• [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration   bilal ghayyad
• [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration   bilal ghayyad
• [asterisk-users] Asterisk and IPTV   bilal ghayyad
• [asterisk-users] Digium IP Phone D40 quality, very bad   bilal ghayyad
• [asterisk-users] after upgrade from 1.4.26.2 to 1.8.11.0 my provider gets rport instead of port   gincantalupo
• [asterisk-users] OT - Integration with building intercom systems   giovanni.v
• [asterisk-users] after upgrade from 1.4.26.2 to 1.8.11.0 my provider gets rport instead of port   giovanni.v
• [asterisk-users] OT - Integration with building intercom systems   giovanni.v
• [asterisk-users] Finding the position of a character in a string   giovanni.v
• [asterisk-users] Asterisk trying to call a queue with no members   isrlgb at gmail.com
• [asterisk-users] Mediatrix 4400plus ISDN setup   hbk
• [asterisk-users] channel not available and jump to next group channels   mahesh katta
• [asterisk-users] channel not available and jump to next group channels   mahesh katta
• [asterisk-users] Outbound Asterisk calls default directmedia specifications   sathiish kumar
• [asterisk-users] Outbound Asterisk calls default directmedia specifications   sathiish kumar
• [asterisk-users] Maximum concurrent calls using call files   sathiish kumar
• [asterisk-users] Maximum concurrent calls using call files   sathiish kumar
• [asterisk-users] wrong RTT QoS information always reported   sathiish kumar
• [asterisk-users] passing arguments to macros from originate command   sathiish kumar
• [asterisk-users] Asterisk 1.8.12 and Fax?   motty.cruz
• [asterisk-users] echo canceler query   motty.cruz
• [asterisk-users] Asterisk Realtime issue after registering withx-lite   motty.cruz
• [asterisk-users] freepbx asterisk   neo nortan
• [asterisk-users] callback on busy   pepesz
• [asterisk-users] callback on busy   pepesz
• [asterisk-users] DAHDI problems   equis software
• [asterisk-users] DAHDI problems   equis software
• [asterisk-users] DAHDI problems   equis software
• [asterisk-users] DAHDI problems   equis software
• [asterisk-users] DAHDI problems   equis software
• [asterisk-users] port 5060 is blocked by ISP   alok srivastava
• [asterisk-users] port 5060 is blocked by ISP   alok srivastava
• [asterisk-users] sip set debug on always showing error   alok srivastava
• [asterisk-users] sip set debug on always showing error   alok srivastava
• [asterisk-users] click to call   alok srivastava
• [asterisk-users] How to Auto Answer a sip phone   upendra
• [asterisk-users] How to Auto Answer a sip phone   upendra
• [asterisk-users] How to Auto Answer a sip phone   upendra
• [asterisk-users] How to Auto Answer a sip phone   upendra
• [asterisk-users] How to set SIP to auto answer in the dial plan .   upendra
• [asterisk-users] How to set SIP to auto answer in the dial plan .   upendra
• [asterisk-users] How to set SIP to auto answer in the dial plan .   upendra
• [asterisk-users] chan_ss7 quick patch to enable RBT   [Digital^Dude] ®
• [asterisk-users] Regrading Speech Recognition.   [Digital^Dude] ®
• [asterisk-users] callback - disa   Федорчук Олег
• [asterisk-users] Multiple DID for SIP Trunk   Федорчук Олег
• [asterisk-users] file and on SayNumber() app   נפתלי מאיר
• [asterisk-users] file and on SayNumber() app   נפתלי מאיר

Last message date: Tue Jul 31 21:35:23 CDT 2012
Archived on: Tue Jul 31 21:48:32 CDT 2012

• Messages sorted by: [ thread ] [ subject ] [ date ]
• More info on this list...