June 2011 Archives by author

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Starting: Wed Jun 1 01:06:02 CDT 2011
Ending: Thu Jun 30 19:42:13 CDT 2011
Messages: 983

• [asterisk-users] chan_mobile one way comunication problem   ing.Achim Alexandru
• [asterisk-users] Asterisk Online Training   Amadu
• [asterisk-users] benefits of asterisk 1.8   Benny Amorsen
• [asterisk-users] Skall vi gå på Smaklösa?   Mattias Andersson
• [asterisk-users] SPA504G Unable to Transfer Established Call   Andres
• [asterisk-users] PAP2T provisioning via SRV record?   Andres
• [asterisk-users] Dahdi_genconfig - "Empty configuration -- no spans"   Sebastian Arcus
• [asterisk-users] [solved] Dahdi_genconfig - "Empty configuration -- no spans"   Sebastian Arcus
• [asterisk-users] Asterisk, attended transfers and DTMF mode   Sebastian Arcus
• [asterisk-users] Cisco IP Phones 7942G (skinny): TFTP and required files   Sebastian Arcus
• [asterisk-users] Problems with pickupgroup/callgroup with Asterisk 1.8.4.2   Sebastian Arcus
• [asterisk-users] Problems with pickupgroup/callgroup with Asterisk 1.8.4.2   Sebastian Arcus
• [asterisk-users] "pickupsound = beep" kills call pickup in Asterisk 1.8.4.2   Sebastian Arcus
• [asterisk-users] "pickupsound = beep" kills call pickup in Asterisk1.8.4.2   Sebastian Arcus
• [asterisk-users] Easier to remember transfer and pick-up key sequences   Sebastian Arcus
• [asterisk-users] AMI buffering event output?   Örn Arnarson
• [asterisk-users] AMI buffering event output?   Örn Arnarson
• [asterisk-users] AMI buffering event output?   Örn Arnarson
• [asterisk-users] AMI buffering event output?   Örn Arnarson
• [asterisk-users] AMI buffering event output?   Örn Arnarson
• [asterisk-users] AMI buffering event output?   Örn Arnarson
• [asterisk-users] test call generator   Daniel - Asterisk
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   Dan Austin
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   Dan Austin
• [asterisk-users] What is wrong in m   Bruce B
• [asterisk-users] Clarification of the terms shown on CLI   Bruce B
• [asterisk-users] t.38 virtual fax software?   David Backeberg
• [asterisk-users] ReceiveFax to G.711   David Backeberg
• [asterisk-users] Asterisk 1.8 minimum modules/configuration   Chris Bagnall
• [asterisk-users] AMI buffering event output?   Alex Balashov
• [asterisk-users] A question about Caller ID   Alex Balashov
• [asterisk-users] Configuring Cisco Phones to register on Asterisk: The configuration files   Alex Balashov
• [asterisk-users] Dial out conference   Alex Balashov
• [asterisk-users] How to secure our Asterisk server from hacker's ?   Alex Balashov
• [asterisk-users] ITSP failover for PRI   Alex Balashov
• [asterisk-users] ITSP failover for PRI   Alex Balashov
• [asterisk-users] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko   Alex Balashov
• [asterisk-users] Conference feature   Alex Balashov
• [asterisk-users] Agi script for working hours PBX   Alex Balashov
• [asterisk-users] No audio format found to offer.   Alex Balashov
• [asterisk-users] Load Balance Trunks   Alex Balashov
• [asterisk-users] DBdeltree: Error deleting key from database   Satish Barot
• [asterisk-users] Does anyone know about asterisk 1.10   Satish Barot
• [asterisk-users] Three-way conference in Asterisk   Satish Barot
• [asterisk-users] benefits of asterisk 1.8   Satish Barot
• [asterisk-users] benefits of asterisk 1.8   Satish Barot
• [asterisk-users] How to continue processing a context after a Hangup   Satish Barot
• [asterisk-users] How to continue processing a context after a Hangup   Satish Barot
• [asterisk-users] benefits of asterisk 1.8   Satish Barot
• [asterisk-users] Asterisk users Calculation   Satish Barot
• [asterisk-users] RealTime Queue Logging in 1.8   Satish Barot
• [asterisk-users] Different callerid for different extensions   Satish Barot
• [asterisk-users] Different callerid for different extensions   Satish Barot
• [asterisk-users] Queue log in MySQL DB   Satish Barot
• [asterisk-users] Queue log in MySQL DB   Satish Barot
• [asterisk-users] How asterisk use pri channel   Satish Barot
• [asterisk-users] Queue not sending call to Agent   Satish Barot
• [asterisk-users] Queue not sending call to Agent   Satish Barot
• [asterisk-users] asterisk queue 'ringall' stratagy   Satish Barot
• [asterisk-users] Have your suggestions on Hardware configuration for Asterisk.   Satish Barot
• [asterisk-users] change destination on digit   Satish Barot
• [asterisk-users] Office timings only work asterisk after that voicemail   Satish Barot
• [asterisk-users] Vm on a System running Asterisk.   Satish Barot
• [asterisk-users] Vm on a System running Asterisk.   Satish Barot
• [asterisk-users] Conference feature   Satish Barot
• [asterisk-users] Agi script for working hours PBX   Satish Barot
• [asterisk-users] SIP Register DOS attack   Paul Belanger
• [asterisk-users] benefits of asterisk 1.8   Paul Belanger
• [asterisk-users] benefits of asterisk 1.8   Paul Belanger
• [asterisk-users] broken SVN asterisk 1.8 ?   Paul Belanger
• [asterisk-users] how to know length of file in seconds   Paul Belanger
• [asterisk-users] After wiki.asterisk.org was upgraded my user no loger exists.   Paul Belanger
• [asterisk-users] After wiki.asterisk.org was upgraded my user no loger exists.   Paul Belanger
• [asterisk-users] Softphone RTP Jitter   Fernando Berretta
• [asterisk-users] Moh on transfer call   Asistar srl Ivan Bolognani
• [asterisk-users] Missed calls and groups   Russell Brown
• [asterisk-users] background audio for inbound leg   Tom Browning
• [asterisk-users] Migration from Mantis to JIRA   Terry Brummell
• [asterisk-users] A question about Caller ID   Terry Brummell
• [asterisk-users] Ground Start ATA / VOIP Gateway   Terry Brummell
• [asterisk-users] Voicemail issue   Terry Brummell
• [asterisk-users] Re : Re : Direct RTP with Asterisk   Terry Brummell
• [asterisk-users] call paging interrupts call when using Mitel 5224   Terry Brummell
• [asterisk-users] call paging interrupts call when using Mitel 5224   Terry Brummell
• [asterisk-users] call paging interrupts call when using Mitel 5224   Terry Brummell
• [asterisk-users] call paging interrupts call when using Mitel 5224   Terry Brummell
• [asterisk-users] call paging interrupts call when using Mitel 5224   Terry Brummell
• [asterisk-users] call paging interrupts call when using Mitel 5224   Terry Brummell
• [asterisk-users] call paging interrupts call when using Mitel 5224   Terry Brummell
• [asterisk-users] call paging interrupts call when using Mitel 5224   Terry Brummell
• [asterisk-users] call paging interrupts call when using Mitel 5224   Terry Brummell
• [asterisk-users] Free CNAM   Pascal Bruno
• [asterisk-users] Migration from Mantis to JIRA   Russell Bryant
• [asterisk-users] Migration from Mantis to JIRA   Russell Bryant
• [asterisk-users] issues.asterisk.org   Russell Bryant
• [asterisk-users] issues.asterisk.org/jira not working   Russell Bryant
• [asterisk-users] unsubscribe   Ross Cameron
• [asterisk-users] No audio after a reinvite changing codec   Matteo Campana
• [asterisk-users] No audio after a reinvite changing codec   Matteo Campana
• [asterisk-users] No audio after a reinvite changing codec   Matteo Campana
• [asterisk-users] No audio after a reinvite changing codec   Matteo Campana
• [asterisk-users] No audio after a reinvite changing codec   Matteo Campana
• [asterisk-users] No audio after a reinvite changing codec ----> with SIP DEBUG!!   Matteo Campana
• [asterisk-users] Pops & clicks at the end of sound files   Neeraj Chand
• [asterisk-users] DTMF issue in app_konference using with asterisk 1.8.3.2   Krishna Sumanth Chava
• [asterisk-users] How to get DTMF in Konference module in Asterisk   Krishna Sumanth Chava
• [asterisk-users] R: How to remove asterisk ?   Carlos Chavez
• [asterisk-users] asterisk voicemail distribution groups   Carlos Chavez
• [asterisk-users] No audio format found to offer.   Carlos Chavez
• [asterisk-users] Asterisk users Calculation   Khaled W. Chehab
• [asterisk-users] Asterisk call limitation   Khaled W. Chehab
• [asterisk-users] Asterisk call limitation   Khaled W. Chehab
• [asterisk-users] Asterisk call limitation   Khaled W. Chehab
• [asterisk-users] Asterisk call limitation   Khaled W. Chehab
• [asterisk-users] Asterisk call limitation   Khaled W. Chehab
• [asterisk-users] Integration of OpenVXI   Adolphe Cher-Aime
• [asterisk-users] Asterisk 1.6.1 Realtime SIP Users   Adolphe Cher-Aime
• [asterisk-users] A question about Caller ID   Christian
• [asterisk-users] A question about Caller ID   Christian
• [asterisk-users] A question about Caller ID   Christian
• [asterisk-users] R: Re: Looking for Email to Fax Solutions   Enrico Cicconi
• [asterisk-users] DTMF begin ignored   Marcelo Ellmann Clemente
• [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel   Tzafrir Cohen
• [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel   Tzafrir Cohen
• [asterisk-users] LXC and Dahdi   Tzafrir Cohen
• [asterisk-users] LXC and Dahdi   Tzafrir Cohen
• [asterisk-users] Dahdi 2.4.0 and Squeeze   Tzafrir Cohen
• [asterisk-users] dahdi_genconf and BRI NT spans in system.conf   Tzafrir Cohen
• [asterisk-users] #include filename   Tzafrir Cohen
• [asterisk-users] Inbound call not dialing exten   Tzafrir Cohen
• [asterisk-users] DIGIUM PRI CARDS REQUIRE   Tzafrir Cohen
• [asterisk-users] How to secure our Asterisk server from hacker's ?   Tzafrir Cohen
• [asterisk-users] Web based call back   Tzafrir Cohen
• [asterisk-users] ooh323 errors while compiling: asterisk 1.8.3 and 1.8.4   Tzafrir Cohen
• [asterisk-users] Blind transfer issue on Asterisk 1.8.4.2   Bart Coninckx
• [asterisk-users] Change to pickups in Asterisk 1.8 - not working on local channels?   David Cunningham
• [asterisk-users] Voice recognition recommendations?   David Cunningham
• [asterisk-users] Asterisk 1.8.4 - Google iCal not working   Matt Darnell
• [asterisk-users] chan_dahdi.c, dtmfmute, rtp.c   David
• [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel   Steve Davies
• [asterisk-users] Refactor of CDR - Comments please.   Steve Davies
• [asterisk-users] Asterisk 1.8 broken MWI   Steve Davies
• [asterisk-users] Asterisk 1.8 broken MWI   Steve Davies
• [asterisk-users] iLBC re-licence   Steve Davies
• [asterisk-users] iLBC re-licence   Steve Davies
• [asterisk-users] Aastra phone # key in dialplan   Steve Davies
• [asterisk-users] Voicemail issue   Alec Davis
• [asterisk-users] "pickupsound = beep" kills call pickup in Asterisk1.8.4.2   Alec Davis
• [asterisk-users] asterisk queue 'ringall' stratagy   Deka, Rajib IN MAA SL
• [asterisk-users] asterisk queue 'ringall' stratagy   Deka, Rajib IN MAA SL
• [asterisk-users] Asterisk changing SIP INFO dtmf duration   Deka, Rajib IN MAA SL
• [asterisk-users] AMI buffering event output?   Mark Deneen
• [asterisk-users] Asterisk GUI - the one from Diguim/Asterisk - issues on Asterisk 1.6x   Mark Deneen
• [asterisk-users] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko   Mark Deneen
• [asterisk-users] issues.asterisk.org/jira not working   Jim Dickenson
• [asterisk-users] issues.asterisk.org/jira not working   Jim Dickenson
• [asterisk-users] background audio for inbound leg   Jim Dickenson
• [asterisk-users] PAP2T provisioning via SRV record?   Mike Diehl
• [asterisk-users] PAP2T provisioning via SRV record?   Mike Diehl
• [asterisk-users] Channel variables not available during xfer?   Mike Diehl
• [asterisk-users] PAP2T provisioning via SRV record?   Mike Diehl
• [asterisk-users] Outgoing calls get dropped on high-latency connections.   Ernie Dunbar
• [asterisk-users] Using Dial() on SIP and DAHDI connections simultaneously   Ernie Dunbar
• [asterisk-users] Outgoing calls get dropped on high-latency connections.   Ernie Dunbar
• [asterisk-users] No audio format found to offer.   Ernie Dunbar
• [asterisk-users] No audio format found to offer.   Ernie Dunbar
• [asterisk-users] Free CNAM   Michelle Dupuis
• [asterisk-users] Aastra phone # key in dialplan   Michelle Dupuis
• [asterisk-users] Aastra phone # key in dialplan   Michelle Dupuis
• [asterisk-users] please help   Camilo Echeverry
• [asterisk-users] MOH uploading is not working with 1.4   Steve Edwards
• [asterisk-users] asterisk 1.6 - 511 Command not permitted causing high CPU usage   Steve Edwards
• [asterisk-users] Asterisk users Calculation   Steve Edwards
• [asterisk-users] AGI STREAM FILE not working?   Steve Edwards
• [asterisk-users] AGI STREAM FILE not working?   Steve Edwards
• [asterisk-users] AGI STREAM FILE not working?   Steve Edwards
• [asterisk-users] Pops & clicks at the end of sound files   Steve Edwards
• [asterisk-users] Pops & clicks at the end of sound files   Steve Edwards
• [asterisk-users] trunk between 2 servers   Steve Edwards
• [asterisk-users] What is wrong in m   Steve Edwards
• [asterisk-users] how asterisk work with VoIP trunk?   Steve Edwards
• [asterisk-users] No IVR listen at device end......SIP phone is working fine   Steve Edwards
• [asterisk-users] how asterisk work with VoIP trunk?   Steve Edwards
• [asterisk-users] Question about voip.ms service.   Steve Edwards
• [asterisk-users] Permanent restart after upgrade   Steve Edwards
• [asterisk-users] TFTP to be installed in Linux same asterisk machine to be used with Cisco   Steve Edwards
• [asterisk-users] call an external number for other server   Steve Edwards
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   Steve Edwards
• [asterisk-users] fast AGI memory leaks   Steve Edwards
• [asterisk-users] Office timings only work asterisk after that voicemail   Steve Edwards
• [asterisk-users] Conference feature   Steve Edwards
• [asterisk-users] Conference feature   Steve Edwards
• [asterisk-users] Load Balance Trunks   Steve Edwards
• [asterisk-users] Realtime Failover - Multiple DSN 's in extconfig.conf   Stuart Elvish
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   Mark Engelhardt
• [asterisk-users] After wiki.asterisk.org was upgraded my user no loger exists.   Jose P. Espinal
• [asterisk-users] After wiki.asterisk.org was upgraded my user no loger exists.   Jose P. Espinal
• [asterisk-users] Configuring Cisco Phones to register on Asterisk: The configuration files   C F
• [asterisk-users] Configuring Cisco Phones to register on Asterisk: The configuration files   C F
• [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel anyway!   C F
• [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel anyway!   C F
• [asterisk-users] DIGIUM PRI CARDS REQUIRE   C F
• [asterisk-users] Conference feature   C F
• [asterisk-users] rtptimeout on 1.8.4   Jon Farmer
• [asterisk-users] Queue Log in Mysql   Henrique Fernandes
• [asterisk-users] Queue Log in Mysql   Henrique Fernandes
• [asterisk-users] PRI issue its BUSY   Cary Fitch
• [asterisk-users] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko   Cary Fitch
• [asterisk-users] benefits of asterisk 1.8   Kevin P. Fleming
• [asterisk-users] benefits of asterisk 1.8   Kevin P. Fleming
• [asterisk-users] issues.asterisk.org/jira not working   Kevin P. Fleming
• [asterisk-users] issues.asterisk.org/jira not working   Kevin P. Fleming
• [asterisk-users] How to set a HA8 board + B400M in NT mode ?   Kevin P. Fleming
• [asterisk-users] Google Voice receiving call problem   Kevin P. Fleming
• [asterisk-users] Asterisk call limitation   Kevin P. Fleming
• [asterisk-users] Problem with detecting fax on PRI/DAHDI channels   Kevin P. Fleming
• [asterisk-users] Problem with detecting fax on PRI/DAHDI channels   Kevin P. Fleming
• [asterisk-users] ReceiveFax to G.711   Kevin P. Fleming
• [asterisk-users] ReceiveFax to G.711   Kevin P. Fleming
• [asterisk-users] IVR   Jose Flores Galicia
• [asterisk-users] Insert name in SIP registry   Jian Gao
• [asterisk-users] Full SIP dial string   Ruddy Gbaguidi
• [asterisk-users] Full SIP dial string   Ruddy Gbaguidi
• [asterisk-users] SMS with Asterisk   Jared Geiger
• [asterisk-users] call transfer back to a sourcing switch   Jerry Geis
• [asterisk-users] Question on how many phones   Jerry Geis
• [asterisk-users] how to tell if call is on hold   Jerry Geis
• [asterisk-users] AMI question   Jerry Geis
• [asterisk-users] AMI question   Jerry Geis
• [asterisk-users] AMI on hold events   Jerry Geis
• [asterisk-users] show channels does not show hold status   Jerry Geis
• [asterisk-users] how to get on hold events with AMI   Jerry Geis
• [asterisk-users] Question on pause in dialing   Jerry Geis
• [asterisk-users] Question on pause in dialing   Jerry Geis
• [asterisk-users] failed to authenticate user message   Jerry Geis
• [asterisk-users] interface to gstreamer   Jerry Geis
• [asterisk-users] asterisk recording problem   Peter Gelencser
• [asterisk-users] example sip.conf for csipsimple?   Chris Gentle
• [asterisk-users] Re : Re : Re : Direct RTP with Asterisk   Lyle Giese
• [asterisk-users] Connect intercom to Asterisk?   Gilles
• [asterisk-users] Connect intercom to Asterisk?   Gilles
• [asterisk-users] Integration of OpenVXI   Jean-Denis Girard
• [asterisk-users] Fax with Asterisk and T38Modem   Thorolf Godawa
• [asterisk-users] SIP/IAX guest access?   Stefan Gofferje
• [asterisk-users] SIP/IAX guest access?   Stefan Gofferje
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   Stefan Gofferje
• [asterisk-users] multiple asterisk on 1 machine or other idea for using multiple network connection   Israel Gottlieb
• [asterisk-users] Looking for Email to Fax Solutions   Paddy Grice
• [asterisk-users] Looking for Email to Fax Solutions   Paddy Grice
• [asterisk-users] PRI hangup request, cause 18   Thorsten Göllner
• [asterisk-users] PRI hangup request, cause 18   Thorsten Göllner
• [asterisk-users] Inbound CallerID displays asterisk   ERIC HERRON
• [asterisk-users] Inbound CallerID displays asterisk   ERIC HERRON
• [asterisk-users] Inbound SMS   ERIC HERRON
• [asterisk-users] Inbound SMS   ERIC HERRON
• [asterisk-users] Asterisk 1.8.4 - Google iCal not working   ERIC HERRON
• [asterisk-users] FW: Asterisk 1.8.4 - Google iCal not working   ERIC HERRON
• [asterisk-users] Incoming Call Recording   Rick Hall
• [asterisk-users] How to secure our Asterisk server from hacker's ?   Faisal Hanif
• [asterisk-users] Vm on a System running Asterisk.   Faisal Hanif
• [asterisk-users] Monitor Asterisk and Ast-gui   Faisal Hanif
• [asterisk-users] Conference feature   Faisal Hanif
• [asterisk-users] not find files in asterisk 1.8   Faisal Hanif
• [asterisk-users] Asterisk 1.8 Paging without DAHDI or MOH Shoutcast with DAHDI   Faisal Hanif
• [asterisk-users] Outgoing calls get dropped on high-latency connections.   Faisal Hanif
• [asterisk-users] Playing with sipvicious ..   Paul Hayes
• [asterisk-users] Connect intercom to Asterisk?   Paul Hayes
• [asterisk-users] PAP2T provisioning via SRV record?   Paul Hayes
• [asterisk-users] Re : Re : Direct RTP with Asterisk   Paul Hayes
• [asterisk-users] ITSP failover for PRI   Claude Hayn
• [asterisk-users] : Re: ITSP failover for PRI   Claude Hayn
• [asterisk-users] standalone PRI-to-SIP converter   Gordon Henderson
• [asterisk-users] Voxbone numbers   Gordon Henderson
• [asterisk-users] Connect intercom to Asterisk?   Gordon Henderson
• [asterisk-users] LXC and Dahdi   Gordon Henderson
• [asterisk-users] Siemens gigaset as180 as a internal mobile extension   Gordon Henderson
• [asterisk-users] LXC and Dahdi   Gordon Henderson
• [asterisk-users] Siemens gigaset as180 as a internal mobile extension   Gordon Henderson
• [asterisk-users] Siemens gigaset as180 as a internal mobile extension   Gordon Henderson
• [asterisk-users] VMX Locator   Darrin Henshaw
• [asterisk-users] VMX Locator   Darrin Henshaw
• [asterisk-users] Problem callerid ignored by using callfiles   Thomas Hoellriegel
• [asterisk-users] Possible timing issue?   Hose
• [asterisk-users] t.38 virtual fax software?   Hose
• [asterisk-users] t.38 virtual fax software?   Hose
• [asterisk-users] for suport server   Akramul Hossain
• [asterisk-users] Question about "null routing" calls to DIDs we don't handle   Steven Howes
• [asterisk-users] Voxbone numbers   Steven Howes
• [asterisk-users] VOICEMAIL CONFIGURATION   Steven Howes
• [asterisk-users] Re : Re : Re : Re : Direct RTP with Asterisk   Steven Howes
• [asterisk-users] Cannot figure out pound key in qwerty keyboard   Steven Howes
• [asterisk-users] Ground Start ATA / VOIP Gateway   Robert Huddleston
• [asterisk-users] [asterisk-biz] Ground Start ATA / VOIP Gateway   Robert Huddleston
• [asterisk-users] Ground Start ATA / VOIP Gateway   Robert Huddleston
• [asterisk-users] Ground Start ATA / VOIP Gateway   Robert Huddleston
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   Robert Huddleston
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   Robert Huddleston
• [asterisk-users] SMS with Asterisk   Robert Huddleston
• [asterisk-users] asterisk queue 'ringall' stratagy   DHAVAL INDRODIYA
• [asterisk-users] call file challenge...   DHAVAL INDRODIYA
• [asterisk-users] SIP Register DOS attack   Ira
• [asterisk-users] Asterisk 1.6 - subscriptions.   Jarek Jarzebowski
• [asterisk-users] Hints problem - NAT problem?   Jarek Jarzebowski
• [asterisk-users] Asterisk - dialog-info+xml - NAT   Jarek Jarzebowski
• [asterisk-users] Asterisk 1.8.4 - Google iCal not working   LL Jason
• [asterisk-users] Asterisk 1.8.4 - Google iCal not working   LL Jason
• [asterisk-users] SMS with Asterisk   Per Jessen
• [asterisk-users] Issues with Digum Repos / AsteriskNOW & Bad Packages   Andrew Joakimsen
• [asterisk-users] Issues with Digum Repos / AsteriskNOW & Bad Packages   Andrew Joakimsen
• [asterisk-users] Asterisk/Kamailio dinner in Madrid thursday next week - June 30th   Olle E Johansson
• [asterisk-users] Connect intercom to Asterisk?   Dan Journo
• [asterisk-users] atxfer fails to read data   Dan Journo
• [asterisk-users] Cannot figure out pound key in qwerty keyboard   Dan Journo
• [asterisk-users] Queue log in MySQL DB   Jonas Kellens
• [asterisk-users] Queue log in MySQL DB   Jonas Kellens
• [asterisk-users] Asterisk and Audiocodes PRI card   Jonas Kellens
• [asterisk-users] Get second cipher in an extension   Jonas Kellens
• [asterisk-users] Call to *2*999... : IP-phone configration   Jonas Kellens
• [asterisk-users] Get second cipher in an extension   Richard Kenner
• [asterisk-users] t.38 virtual fax software?   John Kiniston
• [asterisk-users] issues.asterisk.org   Jeremy Kister
• [asterisk-users] issues.asterisk.org   Jeremy Kister
• [asterisk-users] asterisk + stun   Jeremy Kister
• [asterisk-users] issues/jira   Jeremy Kister
• [asterisk-users] Call to *2*999... : IP-phone configration   Barry L. Kline
• [asterisk-users] Question regarding progressinband   Anatoliy Kounitskiy
• [asterisk-users] xorcom asterisk patch for sending rtp stream to remote oreka server   Marcus Kvarsell
• [asterisk-users] LXC and Dahdi   Jeff LaCoursiere
• [asterisk-users] LXC and Dahdi   Jeff LaCoursiere
• [asterisk-users] Polycom BLF   Jeff LaCoursiere
• [asterisk-users] Siemens gigaset as180 as a internal mobile extension   Jeff LaCoursiere
• [asterisk-users] sccp problem   Pezhman Lali
• [asterisk-users] Different callerid for different extensions   Edwin Lam
• [asterisk-users] Queue not sending call to Agent   Duane Larson
• [asterisk-users] Queue not sending call to Agent   Duane Larson
• [asterisk-users] Queue not sending call to Agent   Duane Larson
• [asterisk-users] Queue not sending call to Agent   Duane Larson
• [asterisk-users] issues.asterisk.org/jira not working   Andrew Latham
• [asterisk-users] issues.asterisk.org/jira not working   Andrew Latham
• [asterisk-users] TFTP to be installed in Linux same asterisk machine to be used with Cisco   Andrew Latham
• [asterisk-users] Dahdi 2.4.0 and Squeeze   Andrew Latham
• [asterisk-users] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko   Andrew Latham
• [asterisk-users] Clarification of the terms shown on CLI   Andrew Latham
• [asterisk-users] Re connecting to SIP Provider with virtual IP, from pacemaker cluster   Cédric Lemarchand
• [asterisk-users] asteriks and ntt Japan   Cristian Leonte
• [asterisk-users] Issues with Digum Repos / AsteriskNOW & Bad Packages   Patrick Lists
• [asterisk-users] iLBC re-licence   Patrick Lists
• [asterisk-users] Call queues on load-balanced asterisks   Thomas Liu
• [asterisk-users] IVR   Ezequiel Lovelle
• [asterisk-users] why doesn't "s" accept incoming call   Doug Lytle
• [asterisk-users] PRI hangup request, cause 18   Doug Lytle
• [asterisk-users] Connected Line ID   Doug Lytle
• [asterisk-users] Callerid issue   Doug Lytle
• [asterisk-users] Connected Line ID   Doug Lytle
• [asterisk-users] TFTP to be installed in Linux same asterisk machine to be used with Cisco   Doug Lytle
• [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel anyway!   Doug Lytle
• [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel   Doug Lytle
• [asterisk-users] Connected Line ID   Doug Lytle
• [asterisk-users] Connected Line ID   Doug Lytle
• [asterisk-users] Cisco IP Phones 7942 and Skinny/SIP in asterisk   Doug Lytle
• [asterisk-users] Cisco IP Phones 7942 and Skinny/SIP in asterisk   Doug Lytle
• [asterisk-users] Asterisk 1.4 func_odbc frustrations   Doug Lytle
• [asterisk-users] [SOLVED] Re: Asterisk 1.4 func_odbc frustrations   Doug Lytle
• [asterisk-users] Asterisk 1.6.1 Realtime SIP Users   Mickael MONSIEUR
• [asterisk-users] Asterisk 1.8.4.2 Now Available (Security Release)   Leif Madsen
• [asterisk-users] MixMonitor   Leif Madsen
• [asterisk-users] MixMonitor   Leif Madsen
• [asterisk-users] Next Asterisk 1.8 Release   Leif Madsen
• [asterisk-users] patch openr2 asterisk 1.4.42   Andre Magalhaes
• [asterisk-users] RealTime Queue Logging in 1.8   Ishfaq Malik
• [asterisk-users] benefits of asterisk 1.8   Ishfaq Malik
• [asterisk-users] RealTime Queue Logging in 1.8   Ishfaq Malik
• [asterisk-users] RealTime Queue Logging in 1.8   Ishfaq Malik
• [asterisk-users] RealTime Queue Logging in 1.8   Ishfaq Malik
• [asterisk-users] Queue Log in Mysql   Ishfaq Malik
• [asterisk-users] AMI Suddenly not giving full response to 'Command'   Ishfaq Malik
• [asterisk-users] Unable to include switch 'Realtime/@' in context   Ishfaq Malik
• [asterisk-users] not find files in asterisk 1.8   Ishfaq Malik
• [asterisk-users] not find files in asterisk 1.8   Ishfaq Malik
• [asterisk-users] asterisk recording problem   Ishfaq Malik
• [asterisk-users] Asked to transmit frame type slin, while native formats is 0x8 (alaw)   Asterisk Man
• [asterisk-users] Asked to transmit frame type slin, while native formats is 0x8 (alaw)   Asterisk Man
• [asterisk-users] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko   Marcelo
• [asterisk-users] standalone PRI-to-SIP converter   Matjaz
• [asterisk-users] Free CNAM   Sherwood McGowan
• [asterisk-users] benefits of asterisk 1.8   Sherwood McGowan
• [asterisk-users] benefits of asterisk 1.8   Sherwood McGowan
• [asterisk-users] benefits of asterisk 1.8   Sherwood McGowan
• [asterisk-users] Asterisk users Calculation   Sherwood McGowan
• [asterisk-users] Asterisk Online Training   Sherwood McGowan
• [asterisk-users] Pops & clicks at the end of sound files   Sherwood McGowan
• [asterisk-users] What is wrong in m   Sherwood McGowan
• [asterisk-users] Looking for Email to Fax Solutions   Anthony Messina
• [asterisk-users] Page() bumps user out of a call   Russ Meyerriecks
• [asterisk-users] Page() bumps user out of a call   Russ Meyerriecks
• [asterisk-users] ReceiveFax to G.711   Michael
• [asterisk-users] ReceiveFax to G.711   Michael
• [asterisk-users] ReceiveFax to G.711   Michael
• [asterisk-users] not find files in asterisk 1.8   Paolo De Michele
• [asterisk-users] not find files in asterisk 1.8   Paolo De Michele
• [asterisk-users] [ resolved ] not find files in asterisk 1.8   Paolo De Michele
• [asterisk-users] DAHDI - skipping sound on voicemail   Mike
• [asterisk-users] asterisk queue 'ringall' stratagy   Mike
• [asterisk-users] Voicemail issue   Mike
• [asterisk-users] MixMonitor - garbled/corrupted WAV files   Mike
• [asterisk-users] MixMonitor - garbled/corrupted WAV files   Mike
• [asterisk-users] MixMonitor - garbled/corrupted WAV files   Mike
• [asterisk-users] MixMonitor - garbled/corrupted WAV files   Mike
• [asterisk-users] Google Voice receiving call problem   Vladimir Mikhelson
• [asterisk-users] Google Voice receiving call problem   Vladimir Mikhelson
• [asterisk-users] t.38 virtual fax software?   Vladimir Mikhelson
• [asterisk-users] broken SVN asterisk 1.8 ?   Barry Miller
• [asterisk-users] Monitor Asterisk and Ast-gui   Flavio Miranda
• [asterisk-users] Conference feature   Flavio Miranda
• [asterisk-users] Add # at the end of dialled number   Flavio Miranda
• [asterisk-users] Add # at the end of dialled number   Flavio Miranda
• [asterisk-users] Connected Line ID   Arjan Kroon | Mobillion
• [asterisk-users] Connected Line ID   Arjan Kroon | Mobillion
• [asterisk-users] Connected Line ID   Arjan Kroon | Mobillion
• [asterisk-users] Connected Line ID   Arjan Kroon | Mobillion
• [asterisk-users] Connected Line ID   Arjan Kroon | Mobillion
• [asterisk-users] Connected Line ID   Arjan Kroon | Mobillion
• [asterisk-users] How to continue processing a context after a Hangup   Antonio Modesto
• [asterisk-users] Asterisk Online Training   Antonio Modesto
• [asterisk-users] Digium WCTDM24XXP DTMF CallerID   Antonio Modesto
• [asterisk-users] Polycom ip320 dtmf issues   Adam Moffett
• [asterisk-users] Access Voicemail Asterisk 1.8 FreeBSD 8.2   Miguel Molina
• [asterisk-users] Permanent restart after upgrade   Larry Moore
• [asterisk-users] No audio after a reinvite changing codec   Larry Moore
• [asterisk-users] No audio after a reinvite changing codec   Larry Moore
• [asterisk-users] Problem with ReceiveFAX app from FFA   Larry Moore
• [asterisk-users] dialplan execution stops after ReceiveFax   Larry Moore
• [asterisk-users] AGI STREAM FILE not working?   Tony Mountifield
• [asterisk-users] How asterisk use pri channel   Richard Mudgett
• [asterisk-users] How asterisk use pri channel   Richard Mudgett
• [asterisk-users] How asterisk use pri channel   Richard Mudgett
• [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel anyway!   Richard Mudgett
• [asterisk-users] CDRs in 1.8   Richard Mudgett
• [asterisk-users] Bridged Digital call   Richard Mudgett
• [asterisk-users] Bridged Digital call   Richard Mudgett
• [asterisk-users] Bridged Digital call   Richard Mudgett
• [asterisk-users] "pickupsound = beep" kills call pickup in Asterisk 1.8.4.2   Richard Mudgett
• [asterisk-users] how to get on hold events with AMI   Richard Mudgett
• [asterisk-users] How to set BRI-to-BRI trunk using 2 HA8+B400M cards   Richard Mudgett
• [asterisk-users] How to set BRI-to-BRI trunk using 2 HA8+B400M cards   Richard Mudgett
• [asterisk-users] Problem with detecting fax on PRI/DAHDI channels   Richard Mudgett
• [asterisk-users] Asterisk 1.6 Dahdi on Centos 5.2   Richard Mudgett
• [asterisk-users] How to secure our Asterisk server from hacker's ?   RSCL Mumbai
• [asterisk-users] Immediate 180 Response on Invite   Elliot Murdock
• [asterisk-users] Google Voice receiving call problem   Elliot Murdock
• [asterisk-users] Google Voice receiving call problem   Elliot Murdock
• [asterisk-users] Google Voice receiving call problem   Elliot Murdock
• [asterisk-users] Google Voice receiving call problem   Elliot Murdock
• [asterisk-users] Incoming Call Recording   Danny Nicholas
• [asterisk-users] MixMonitor   Danny Nicholas
• [asterisk-users] MixMonitor   Danny Nicholas
• [asterisk-users] menu issue   Danny Nicholas
• [asterisk-users] Add # at the end of dialled number   Danny Nicholas
• [asterisk-users] IVR   Danny Nicholas
• [asterisk-users] Cannot figure out pound key in qwerty keyboard   Danny Nicholas
• [asterisk-users] Cannot figure out pound key in qwerty keyboard   Danny Nicholas
• [asterisk-users] Three-way conference in Asterisk   Nikhil
• [asterisk-users] Does anyone know about asterisk 1.10   Nikhil
• [asterisk-users] MOH uploading is not working with 1.4   Nikhil
• [asterisk-users] MOH uploading is not working with 1.4   Nikhil
• [asterisk-users] MOH uploading is not working with 1.4   Nikhil
• [asterisk-users] Dial out conference   Nikhil
• [asterisk-users] Dial out conference   Nikhil
• [asterisk-users] different format in asterisk   Nikhil
• [asterisk-users] Free CNAM   John Novack
• [asterisk-users] Question about voip.ms service.   John Novack
• [asterisk-users] Ground Start ATA / VOIP Gateway   John Novack
• [asterisk-users] Ground Start ATA / VOIP Gateway   John Novack
• [asterisk-users] Ground Start ATA / VOIP Gateway   John Novack
• [asterisk-users] Ground Start ATA / VOIP Gateway   John Novack
• [asterisk-users] Conference feature   John Novack
• [asterisk-users] LXC and Dahdi   Olivier
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   Olivier
• [asterisk-users] Dahdi 2.4.0 and Squeeze   Olivier
• [asterisk-users] Dahdi 2.4.0 and Squeeze [SOLVED]   Olivier
• [asterisk-users] How to set a HA8 board + B400M in NT mode ?   Olivier
• [asterisk-users] How asterisk use pri channel   Olivier
• [asterisk-users] How to set a HA8 board + B400M in NT mode ? [SOLVED]   Olivier
• [asterisk-users] dahdi_genconf and BRI NT spans in system.conf   Olivier
• [asterisk-users] How to configure unused BRI ports from a HA8 board ?   Olivier
• [asterisk-users] How to configure unused BRI ports from a HA8 board ? [SOLVED]   Olivier
• [asterisk-users] dahdi_genconf and BRI NT spans in system.conf   Olivier
• [asterisk-users] ooh323 errors while compiling: asterisk 1.8.3 and 1.8.4   Olivier
• [asterisk-users] Direct RTP with Asterisk   Olivier
• [asterisk-users] ITSP failover for PRI   Olivier
• [asterisk-users] ITSP failover for PRI   Olivier
• [asterisk-users] Digium HA8. Does settings order matters (in system.conf) ?   Olivier
• [asterisk-users] How to set BRI-to-BRI trunk using 2 HA8+B400M cards   Olivier
• [asterisk-users] How to set BRI-to-BRI trunk using 2 HA8+B400M cards   Olivier
• [asterisk-users] How to set BRI-to-BRI trunk using 2 HA8+B400M cards   Olivier
• [asterisk-users] Digium HA8. Does settings order matters (in system.conf) ?   Olivier
• [asterisk-users] How to set BRI-to-BRI trunk using 2 HA8+B400M cards [SOLVED]   Olivier
• [asterisk-users] Set a specific BLF key on Polycom 650   Olivier
• [asterisk-users] Polycom BLF   Olivier
• [asterisk-users] Set a specific BLF key on Polycom 650 [SOLVED]   Olivier
• [asterisk-users] OT - Polycom - Which provisioning protocol to choose ?   Olivier
• [asterisk-users] OT - Polycom - Which provisioning protocol to choose ?   Olivier
• [asterisk-users] AMI buffering event output?   Olivier
• [asterisk-users] R: How to remove asterisk ?   Alexandru Oniciuc
• [asterisk-users] cisco sip   Kelly Opal
• [asterisk-users] call file challenge...   Positively Optimistic
• [asterisk-users] call file challenge...   Positively Optimistic
• [asterisk-users] benefits of asterisk 1.8   Chris Owen
• [asterisk-users] Asked to transmit frame type slin, while native formats is 0x8 (alaw)   Fellipe Paes
• [asterisk-users] Google Voice receiving call problem   Tim Panton
• [asterisk-users] Google Voice receiving call problem   Tim Panton
• [asterisk-users] Monitor Asterisk and Ast-gui   Tim Panton
• [asterisk-users] benefits of asterisk 1.8   Satish Patel
• [asterisk-users] broken SVN asterisk 1.8 ?   Satish Patel
• [asterisk-users] asterisk 1.8 issue with polycom dialplan   Satish Patel
• [asterisk-users] PRI hangup request, cause 18   Satish Patel
• [asterisk-users] issues.asterisk.org/jira not working   Satish Patel
• [asterisk-users] issues.asterisk.org/jira not working   Satish Patel
• [asterisk-users] How asterisk use pri channel   Satish Patel
• [asterisk-users] How asterisk use pri channel   Satish Patel
• [asterisk-users] issues.asterisk.org/jira not working   Satish Patel
• [asterisk-users] Interesting PRI issue   Satish Patel
• [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1 No D-channels available! Using Primary channel as D-channel anyway!   Satish Patel
• [asterisk-users] Asterisk call limitation   Satish Patel
• [asterisk-users] Asterisk call limitation   Satish Patel
• [asterisk-users] PAP2T provisioning via SRV record?   Marius Pedersen
• [asterisk-users] PAP2T provisioning via SRV record?   Marius Pedersen
• [asterisk-users] OT - Polycom - Which provisioning protocol to choose ?   Marius Pedersen
• [asterisk-users] Vm on a System running Asterisk.   Perenaster
• [asterisk-users] Asterisk 1.8 Paging without DAHDI or MOH Shoutcast with DAHDI   Bob Pierce
• [asterisk-users] AMI buffering event output?   Matt Riddell
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   Robert-iPhone
• [asterisk-users] Ground Start ATA / VOIP Gateway   Robert-iPhone
• [asterisk-users] Ground Start ATA / VOIP Gateway   Robert-iPhone
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   Robert-iPhone
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   Robert-iPhone
• [asterisk-users] t.38 virtual fax software?   Robert-iPhone
• [asterisk-users] tls/srtp: sip_xmit error: returned -2   Da Rock
• [asterisk-users] tls/srtp: sip_xmit error: returned -2   Da Rock
• [asterisk-users] HELP! tls/srtp: sip_xmit error: returned -2   Da Rock
• [asterisk-users] Direct RTP with Asterisk   Sagbo Romaric
• [asterisk-users] Re : Direct RTP with Asterisk   Sagbo Romaric
• [asterisk-users] Re : Re : Direct RTP with Asterisk   Sagbo Romaric
• [asterisk-users] Re : Re : Re : Direct RTP with Asterisk   Sagbo Romaric
• [asterisk-users] Re : Re : Re : Direct RTP with Asterisk   Sagbo Romaric
• [asterisk-users] Re : Re : Re : Re : Direct RTP with Asterisk   Sagbo Romaric
• [asterisk-users] Re connecting to SIP Provider with virtual IP, from pacemaker cluster   Torsten Rosenberger
• [asterisk-users] Digium HA8. Does settings order matters (in system.conf) ?   Shaun Ruffell
• [asterisk-users] MixMonitor - garbled/corrupted WAV files   Shaun Ruffell
• [asterisk-users] MixMonitor - garbled/corrupted WAV files   Shaun Ruffell
• [asterisk-users] dialplan execution stops after ReceiveFax   Ruben Rögels
• [asterisk-users] OT - Polycom - Which provisioning protocol to choose ?   Ruben Rögels
• [asterisk-users] dialplan execution stops after ReceiveFax   Ruben Rögels
• [asterisk-users] SendFax: not setting the fax header   Ruben Rögels
• [asterisk-users] Failover trunks   Abid Saleem
• [asterisk-users] Load Balance Trunks   Abid Saleem
• [asterisk-users] calleridname presentation Asterisk ==> Siemens   Rafael dos Santos Saraiva
• [asterisk-users] Conference feature   Rafael dos Santos Saraiva
• [asterisk-users] Conference feature   Rafael dos Santos Saraiva
• [asterisk-users] How to continue processing a context after a Hangup   Warren Selby
• [asterisk-users] Question about "null routing" calls to DIDs we don't handle   Warren Selby
• [asterisk-users] Callerid issue   Warren Selby
• [asterisk-users] Callerid issue   Warren Selby
• [asterisk-users] Asterisk issue or VoIP provider issue ?   Warren Selby
• [asterisk-users] asterisk 1.8 PRI random call drop issue   Warren Selby
• [asterisk-users] Goggle voice incoming dialplan   Warren Selby
• [asterisk-users] Goggle voice incoming dialplan   Warren Selby
• [asterisk-users] asterisk voicemail distribution groups   Warren Selby
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   Warren Selby
• [asterisk-users] SMS with Asterisk   Warren Selby
• [asterisk-users] Problem with ReceiveFAX app from FFA   Warren Selby
• [asterisk-users] menu issue   Warren Selby
• [asterisk-users] Get second cipher in an extension   Warren Selby
• [asterisk-users] Problem with ReceiveFAX app from FFA   Warren Selby
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   Warren Selby
• [asterisk-users] Inbound CallerID displays asterisk   Warren Selby
• [asterisk-users] : Re: ITSP failover for PRI   Warren Selby
• [asterisk-users] SMS with Asterisk   Warren Selby
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   Warren Selby
• [asterisk-users] MixMonitor - garbled/corrupted WAV files   Warren Selby
• [asterisk-users] Set a specific BLF key on Polycom 650   Warren Selby
• [asterisk-users] Asterisk 1.6 Dahdi on Centos 5.2   Warren Selby
• [asterisk-users] Fwd: FW: loadstar and ok one   Vidura Senadeera
• [asterisk-users] Help launch the stackexchange (stackoverflow) telophony site   Jacinda Shelly
• [asterisk-users] sipp application/dtmf-relay not work properly in Asterisk!   Zhang Shukun
• [asterisk-users] standalone PRI-to-SIP converter   Andreas Sikkema
• [asterisk-users] Re : Re : Re : Direct RTP with Asterisk   Andreas Sikkema
• [asterisk-users] Free CNAM   Skyler
• [asterisk-users] Free CNAM   Skyler
• [asterisk-users] Pops & clicks at the end of sound files   Skyler
• [asterisk-users] Page() bumps user out of a call   Cassius Smith
• [asterisk-users] Page() bumps user out of a call   Cassius Smith
• [asterisk-users] Cisco IP Phones 7942G (skinny): TFTP and required files   Cassius Smith
• [asterisk-users] dropped calls on android voip connection   Eric Smith
• [asterisk-users] Asterisk/SIP Issue - Long Shot   Marc Smith
• [asterisk-users] SIP/IAX guest access?   Jamie A. Stapleton
• [asterisk-users] SIP/IAX guest access?   Jamie A. Stapleton
• [asterisk-users] Asterisk issue or VoIP provider issue ?   Jamie A. Stapleton
• [asterisk-users] Goggle voice incoming dialplan   Jamie A. Stapleton
• [asterisk-users] Can I use phone line to recive faxes?   A J Stiles
• [asterisk-users] Callerid issue   A J Stiles
• [asterisk-users] Callerid issue   A J Stiles
• [asterisk-users] Callerid issue   A J Stiles
• [asterisk-users] How to configure unused BRI ports from a HA8 board ?   A J Stiles
• [asterisk-users] Inbound call not dialing exten   A J Stiles
• [asterisk-users] Inbound call not dialing exten   A J Stiles
• [asterisk-users] Load Balance Trunks   A J Stiles
• [asterisk-users] Asterisk/SIP Issue - Long Shot   A J Stiles
• [asterisk-users] Cannot figure out pound key in qwerty keyboard   A J Stiles
• [asterisk-users] issues.asterisk.org/jira not working   William Stillwell
• [asterisk-users] Full SIP dial string   William Stillwell
• [asterisk-users] Goggle voice incoming dialplan   William Stillwell
• [asterisk-users] Goggle voice incoming dialplan   William Stillwell
• [asterisk-users] Setting the Source IP on asterisk 1.8   Davis Sylvester
• [asterisk-users] IRC canal   Florent THOMAS
• [asterisk-users] [FreePBX] Digium addons   Florent THOMAS
• [asterisk-users] How to remove asterisk ?   Florent THOMAS
• [asterisk-users] [FreePBX] Digium addons   Florent THOMAS
• [asterisk-users] Siemens gigaset as180 as a internal mobile extension   Florent THOMAS
• [asterisk-users] vTiger component   Florent THOMAS
• [asterisk-users] Siemens gigaset as180 as a internal mobile extension   Florent THOMAS
• [asterisk-users] Communciation delay betwwn speakers   Florent THOMAS
• [asterisk-users] announce-frequency not respected   Florent THOMAS
• [asterisk-users] Communciation delay betwwn speakers   Florent THOMAS
• [asterisk-users] Communciation delay betwwn speakers   Florent THOMAS
• [asterisk-users] Siemens gigaset as180 as a internal mobile extension   Florent THOMAS
• [asterisk-users] Siemens gigaset as180 as a internal mobile extension   Florent THOMAS
• [asterisk-users] Siemens gigaset as180 as a internal mobile extension   Florent THOMAS
• [asterisk-users] Siemens gigaset as180 as a internal mobile extension   Florent THOMAS
• [asterisk-users] Siemens gigaset as180 as a internal mobile extension   Florent THOMAS
• [asterisk-users] Siemens gigaset as180 as a internal mobile extension   Florent THOMAS
• [asterisk-users] Asterisk 1.4.41 - Warning and Notice about contact info and stale nonce   Administrator TOOTAI
• [asterisk-users] Inbound SMS   Administrator TOOTAI
• [asterisk-users] Asterisk 1.8.4.2 Now Available (Security Release)   Asterisk Development Team
• [asterisk-users] Asterisk 1.4.41.1, 1.6.2.18.1, and 1.8.4.3 Now Available (Security Release)   Asterisk Development Team
• [asterisk-users] Asterisk 1.4.41.2, 1.6.2.18.2, and 1.8.4.4 Now Available (Security Releases)   Asterisk Development Team
• [asterisk-users] Asterisk 1.4.42 Now Available (Final Maintenance Release)   Asterisk Development Team
• [asterisk-users] Asterisk 1.6.2.19 Now Available (Final Maintenance Release)   Asterisk Development Team
• [asterisk-users] AST-2011-011: Possible enumeration of SIP users due to differing authentication responses   Asterisk Security Team
• [asterisk-users] Obtain SIP From and To Tag for CDR   Robert Thomas
• [asterisk-users] HDLC Overrun with Chan SS7   Robert Thomas
• [asterisk-users] Question about "null routing" calls to DIDs we don't handle   Jesse Thompson
• [asterisk-users] asterisk-users Digest, Vol 83, Issue 3   Jesse Thompson
• [asterisk-users] Question about "null routing" calls to DIDs we don't handle   Jesse Thompson
• [asterisk-users] Asterisk GUI - the one from Diguim/Asterisk - issues on Asterisk 1.6x   Silver Thorne
• [asterisk-users] Question about voip.ms service.   Silver Thorne
• [asterisk-users] Google Voice receiving call problem   Silver Thorne
• [asterisk-users] IPv6 and IPv4 NAT not working   Christoph Timm
• [asterisk-users] Update problem | CLI commands missing   Christoph Timm
• [asterisk-users] Update problem | CLI commands missing   Christoph Timm
• [asterisk-users] Asterisk Online Training   Steve Totaro
• [asterisk-users] Permanent restart after upgrade   Steve Totaro
• [asterisk-users] Permanent restart after upgrade   Steve Totaro
• [asterisk-users] Callerid issue   Steve Totaro
• [asterisk-users] Callerid issue   Steve Totaro
• [asterisk-users] TFTP to be installed in Linux same asterisk machine to be used with Cisco   Steve Totaro
• [asterisk-users] TFTP to be installed in Linux same asterisk machine to be used with Cisco   Steve Totaro
• [asterisk-users] SMS with Asterisk   Steve Totaro
• [asterisk-users] SMS with Asterisk   Steve Totaro
• [asterisk-users] SMS with Asterisk   Steve Totaro
• [asterisk-users] SMS with Asterisk   Steve Totaro
• [asterisk-users] Problem with ReceiveFAX app from FFA   Steve Underwood
• [asterisk-users] Problem with ReceiveFAX app from FFA   Steve Underwood
• [asterisk-users] Asterisk DTMF 'talkoff' issues   Steve Underwood
• [asterisk-users] Polycom BLF   Gord Urquhart
• [asterisk-users] Polycom BLF   Gord Urquhart
• [asterisk-users] No IVR listen at device end......SIP phone is working fine   RAJNIKANT VANZA
• [asterisk-users] Office timings only work asterisk after that voicemail   RAJNIKANT VANZA
• [asterisk-users] Looking for Sipura-2000 Latest Firmware   Amol Vedak
• [asterisk-users] Looking for Sipura-2000 Latest Firmware   Amol Vedak
• [asterisk-users] Request: please test modification to EWS calendar functionality   Michel Verbraak
• [asterisk-users] Request: please test modification to EWS calendar functionality   Michel Verbraak
• [asterisk-users] Asterisk: BYE is received late   Vieri
• [asterisk-users] Asterisk: BYE is received late   Vieri
• [asterisk-users] ChannelRedirect   Alex Vishnev
• [asterisk-users] Bridged Call   Alex Vishnev
• [asterisk-users] Problem with detecting fax on PRI/DAHDI channels   Sawan Vithlani
• [asterisk-users] Problem with detecting fax on PRI/DAHDI channels   Sawan Vithlani
• [asterisk-users] Problem with detecting fax on PRI/DAHDI channels   Sawan Vithlani
• [asterisk-users] Problem with detecting fax on PRI/DAHDI channels   Sawan Vithlani
• [asterisk-users] trunk between 2 servers   Robin Vleij
• [asterisk-users] Connected Line ID   Ryan Wagoner
• [asterisk-users] VMX Locator   Ryan Wagoner
• [asterisk-users] why doesn't "s" accept incoming call   Chad Wallace
• [asterisk-users] Free CNAM   Michael R. Wally
• [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel   Karsten Wemheuer
• [asterisk-users] how to know length of file in seconds   Karsten Wemheuer
• [asterisk-users] Voicemail issue   Karsten Wemheuer
• [asterisk-users] click to call   Roger Burton West
• [asterisk-users] Re : Direct RTP with Asterisk   Roger Burton West
• [asterisk-users] benefits of asterisk 1.8   Eric Wieling
• [asterisk-users] benefits of asterisk 1.8   Eric Wieling
• [asterisk-users] why doesn't "s" accept incoming call   Eric Wieling
• [asterisk-users] reload chan_dahdi.conf without disconnect active calls   Eric Wieling
• [asterisk-users] Asterisk 1.8 broken MWI   Eric Wieling
• [asterisk-users] Asterisk 1.8 broken MWI   Eric Wieling
• [asterisk-users] Asterisk 1.8 broken MWI   Eric Wieling
• [asterisk-users] Asterisk 1.8 broken MWI   Eric Wieling
• [asterisk-users] AMI question   Eric Wieling
• [asterisk-users] AMI question   Eric Wieling
• [asterisk-users] AMI on hold events   Eric Wieling
• [asterisk-users] Communciation delay betwwn speakers   Eric Wieling
• [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1: D-chanannel anyway!   Eric Wieling
• [asterisk-users] Voicemail issue   Eric Wieling
• [asterisk-users] Voicemail issue   Eric Wieling
• [asterisk-users] No audio after a reinvite changing codec   Eric Wieling
• [asterisk-users] asterisk voicemail distribution groups   Eric Wieling
• [asterisk-users] No audio after a reinvite changing codec   Eric Wieling
• [asterisk-users] Connected Line ID   Eric Wieling
• [asterisk-users] Connected Line ID   Eric Wieling
• [asterisk-users] Re : Re : Direct RTP with Asterisk   Eric Wieling
• [asterisk-users] Re : Re : Re : Direct RTP with Asterisk   Eric Wieling
• [asterisk-users] Get second cipher in an extension   Eric Wieling
• [asterisk-users] Inbound SMS   Eric Wieling
• [asterisk-users] Inbound SMS   Eric Wieling
• [asterisk-users] Question on pause in dialing   Eric Wieling
• [asterisk-users] Question on pause in dialing   Eric Wieling
• [asterisk-users] Problem with detecting fax on PRI/DAHDI channels   Eric Wieling
• [asterisk-users] Problem with detecting fax on PRI/DAHDI channels   Eric Wieling
• [asterisk-users] Problem with detecting fax on PRI/DAHDI channels   Eric Wieling
• [asterisk-users] Cannot figure out pound key in qwerty keyboard   Eric Wieling
• [asterisk-users] Cannot figure out pound key in qwerty keyboard   Eric Wieling
• [asterisk-users] No audio format found to offer.   Eric Wieling
• [asterisk-users] ControlPlayback's options   Johan Wilfer
• [asterisk-users] Fwd: Re: ControlPlayback's options   Johan Wilfer
• [asterisk-users] Fwd: Re: ControlPlayback's options   Johan Wilfer
• [asterisk-users] [asterisk-biz] Ground Start ATA / VOIP Gateway   Mark Willis
• [asterisk-users] sysmon on Centos Asterisk system using 100 perc CPU..How to kill it?????   Shaun Wingrin
• [asterisk-users] benefits of asterisk 1.8   Hans Witvliet
• [asterisk-users] half sip registration at 1.8.3   Hans Witvliet
• [asterisk-users] Permanent restart after upgrade   Hans Witvliet
• [asterisk-users] Permanent restart after upgrade   Hans Witvliet
• [asterisk-users] Permanent restart after upgrade   Hans Witvliet
• [asterisk-users] Solved, was: Permanent restart after upgrade   Hans Witvliet
• [asterisk-users] Cisco IP Phones 7942G (skinny): TFTP and required files   Ian S. Worthington
• [asterisk-users] About Asterisk SIP NAT Config   Koichi Yagishita
• [asterisk-users] Asterisk 1.8 minimum modules/configuration   Lefteris Zafiris
• [asterisk-users] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko   Norbert Zawodsky
• [asterisk-users] Voicemail issue   Bryant Zimmerman
• [asterisk-users] Voicemail issue   Bryant Zimmerman
• [asterisk-users] Problem with ReceiveFAX app from FFA   Bryant Zimmerman
• [asterisk-users] Problem with ReceiveFAX app from FFA   Bryant Zimmerman
• [asterisk-users] Monitor Asterisk and Ast-gui   Richard Zulu
• [asterisk-users] AGI STREAM FILE not working?   A E [Gmail]
• [asterisk-users] AGI STREAM FILE not working?   A E [Gmail]
• [asterisk-users] AGI STREAM FILE not working?   A E [Gmail]
• [asterisk-users] AGI STREAM FILE not working?   A E [Gmail]
• [asterisk-users] AGI STREAM FILE not working?   A E [Gmail]
• [asterisk-users] AGI STREAM FILE not working?   A E [Gmail]
• [asterisk-users] Subject Change: Playback/background App over Network (was: AGI STREAM FILE not working?)   A E [Gmail]
• [asterisk-users] Next Asterisk 1.8 Release   --[ UxBoD ]--
• [asterisk-users] (no subject)   fabio alves
• [asterisk-users] Polycom ip320 dtmf issues   vmedina at apcn.net
• [asterisk-users] Polycom ip320 dtmf issues   vmedina at apcn.net
• [asterisk-users] Polycom ip320 dtmf issues   vmedina at apcn.net
• [asterisk-users] Goggle voice incoming dialplan   asterisk asterisk
• [asterisk-users] Web based call back   asterisk asterisk
• [asterisk-users] Goggle voice incoming dialplan   asterisk asterisk
• [asterisk-users] Goggle voice incoming dialplan   asterisk asterisk
• [asterisk-users] Goggle voice incoming dialplan   asterisk asterisk
• [asterisk-users] Web based call back   asterisk asterisk
• [asterisk-users] Goggle voice incoming dialplan   asterisk asterisk
• [asterisk-users] Is this Asterisk issue of feature   virendra bhati
• [asterisk-users] ControlPlayback's options   virendra bhati
• [asterisk-users] ControlPlayback's options   virendra bhati
• [asterisk-users] How to get DTMF in Konference module in Asterisk   virendra bhati
• [asterisk-users] Is this feature or Bug of all Asterisk versions ?   virendra bhati
• [asterisk-users] how to know length of file in seconds   virendra bhati
• [asterisk-users] how to know length of file in seconds   virendra bhati
• [asterisk-users] How to get DTMF in Konference module in Asterisk   virendra bhati
• [asterisk-users] how to know length of file in seconds   virendra bhati
• [asterisk-users] how asterisk work with VoIP trunk?   virendra bhati
• [asterisk-users] CallerID issue   virendra bhati
• [asterisk-users] No IVR listen at device end......SIP phone is working fine   virendra bhati
• [asterisk-users] how asterisk work with VoIP trunk?   virendra bhati
• [asterisk-users] No IVR listen at device end......SIP phone is working fine   virendra bhati
• [asterisk-users] Fwd: Re: ControlPlayback's options   virendra bhati
• [asterisk-users] How to remove asterisk ?   virendra bhati
• [asterisk-users] Asterisk issue or VoIP provider issue ?   virendra bhati
• [asterisk-users] R: How to remove asterisk ?   virendra bhati
• [asterisk-users] Dial out conference   virendra bhati
• [asterisk-users] DIGIUM PRI CARDS REQUIRE   virendra bhati
• [asterisk-users] change destination on digit   virendra bhati
• [asterisk-users] How to secure our Asterisk server from hacker's ?   virendra bhati
• [asterisk-users] How to secure our Asterisk server from hacker's ?   virendra bhati
• [asterisk-users] Sending SMS on Friday at 12 Noon EDT   virendra bhati
• [asterisk-users] How to secure our Asterisk server from hacker's ?   virendra bhati
• [asterisk-users] SMS with Asterisk   virendra bhati
• [asterisk-users] SMS with Asterisk   virendra bhati
• [asterisk-users] Asterisk DTMF 'talkoff' issues   virendra bhati
• [asterisk-users] How to know how many calls are into hold byasterisk command   virendra bhati
• [asterisk-users] Bridged Digital call   robert boardman
• [asterisk-users] CDRs in 1.8   robert boardman
• [asterisk-users] Bridged Digital call   robert boardman
• [asterisk-users] Bridged Digital call   robert boardman
• [asterisk-users] Google Voice receiving call problem   cobra2
• [asterisk-users] Goggle voice incoming dialplan   cobra2
• [asterisk-users] example sip.conf for csipsimple?   sean darcy
• [asterisk-users] why doesn't "s" accept incoming call   sean darcy
• [asterisk-users] Voxbone numbers   devr devr
• [asterisk-users] Voxbone numbers   devr devr
• [asterisk-users] Voxbone numbers   devr devr
• [asterisk-users] benefits of asterisk 1.8   dotnetdub
• [asterisk-users] please help   salaheddine elharit
• [asterisk-users] trunk between 2 servers   salaheddine elharit
• [asterisk-users] call an external number for other server   salaheddine elharit
• [asterisk-users] MixMonitor   salaheddine elharit
• [asterisk-users] MixMonitor   salaheddine elharit
• [asterisk-users] MixMonitor   salaheddine elharit
• [asterisk-users] MixMonitor   salaheddine elharit
• [asterisk-users] click to call   salaheddine elharit
• [asterisk-users] menu issue   salaheddine elharit
• [asterisk-users] menu issue   salaheddine elharit
• [asterisk-users] menu issue   salaheddine elharit
• [asterisk-users] menu issue   salaheddine elharit
• [asterisk-users] menu asterisk   salaheddine elharit
• [asterisk-users] asterisk recording problem   salaheddine elharit
• [asterisk-users] TFTP to be installed in Linux same asterisk machine to be used with Cisco   bilal ghayyad
• [asterisk-users] TFTP to be installed in Linux same asterisk machine to be used with Cisco   bilal ghayyad
• [asterisk-users] Configuring Cisco Phones to register on Asterisk: The configuration files   bilal ghayyad
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   bilal ghayyad
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   bilal ghayyad
• [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel anyway!   bilal ghayyad
• [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel   bilal ghayyad
• [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel anyway!   bilal ghayyad
• [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1: D-chanannel anyway!   bilal ghayyad
• [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel anyway!   bilal ghayyad
• [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel anyway!   bilal ghayyad
• [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1 No D-channels available! Using Primary channel as D-channel anyway!   bilal ghayyad
• [asterisk-users] sig_pri.c:985 pri_find_dchan: Span 1 No D-channels available! Using Primary channel as D-channel anyway!   bilal ghayyad
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   bilal ghayyad
• [asterisk-users] Cisco IP Phones 7942G (skinny): TFTP and required files   bilal ghayyad
• [asterisk-users] ooh323 errors while compiling: asterisk 1.8.3 and 1.8.4   bilal ghayyad
• [asterisk-users] ooh323 errors while compiling: asterisk 1.8.3 and 1.8.4   bilal ghayyad
• [asterisk-users] ooh323 errors while compiling asterisk 1.8.3 and 1.8.4   bilal ghayyad
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   bilal ghayyad
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   bilal ghayyad
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   bilal ghayyad
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   bilal ghayyad
• [asterisk-users] Cisco IP Phones and Skinny in asterisk   bilal ghayyad
• [asterisk-users] Cisco IP Phones and Skinny in asterisk 1.8.4.2 "tooooooooooooooooo"   bilal ghayyad
• [asterisk-users] Cisco IP Phones 7942 and Skinny/SIP in asterisk   bilal ghayyad
• [asterisk-users] Cisco IP Phones 7942 and Skinny/SIP in asterisk   bilal ghayyad
• [asterisk-users] chan-mobile bug: bluetooth connection is disconnected immediately when the call hangup   huu giang
• [asterisk-users] Dahdi 2.4.0 and Squeeze [SOLVED]   gincantalupo
• [asterisk-users] Queue not sending call to Agent   duane.larson at gmail.com
• [asterisk-users] Callerid issue   isrlgb at gmail.com
• [asterisk-users] Missed calls and groups   isrlgb at gmail.com
• [asterisk-users] ReceiveFax to G.711   isrlgb at gmail.com
• [asterisk-users] Playing with sipvicious ..   hh174
• [asterisk-users] Voxbone numbers   hh174
• [asterisk-users] Aastra phone # key in dialplan   marvin horst
• [asterisk-users] Aastra phone # key in dialplan   marvin horst
• [asterisk-users] Cannot figure out pound key in qwerty keyboard   michael k
• [asterisk-users] Cannot figure out pound key in qwerty keyboard   michael k
• [asterisk-users] Cannot figure out pound key in qwerty keyboard   michael k
• [asterisk-users] Cannot figure out pound key in qwerty keyboard   michael k
• [asterisk-users] Cannot figure out pound key in qwerty keyboard   michael k
• [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel   mahesh katta
• [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel   mahesh katta
• [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel   mahesh katta
• [asterisk-users] AGI STREAM FILE not working?   mahesh katta
• [asterisk-users] Different callerid for different extensions   mahesh katta
• [asterisk-users] Different callerid for different extensions   mahesh katta
• [asterisk-users] Call Conference and Call transfer and Voice mail settings   mahesh katta
• [asterisk-users] Different callerid for different extensions   mahesh katta
• [asterisk-users] Callerid issue   mahesh katta
• [asterisk-users] Callerid issue   mahesh katta
• [asterisk-users] Callerid issue   mahesh katta
• [asterisk-users] Callerid issue   mahesh katta
• [asterisk-users] VOICEMAIL CONFIGURATION   mahesh katta
• [asterisk-users] VOICEMAIL CONFIGURATION   mahesh katta
• [asterisk-users] CONFERENCE CONFIGURATION REQUIRE   mahesh katta
• [asterisk-users] DIGIUM PRI CARDS REQUIRE   mahesh katta
• [asterisk-users] DIGIUM PRI CARDS REQUIRE   mahesh katta
• [asterisk-users] #include filename   mahesh katta
• [asterisk-users] Inbound call not dialing exten   mahesh katta
• [asterisk-users] Inbound call not dialing exten   mahesh katta
• [asterisk-users] Inbound call not dialing exten   mahesh katta
• [asterisk-users] Office timings only work asterisk after that voicemail   mahesh katta
• [asterisk-users] Office timings only work asterisk after that voicemail   mahesh katta
• [asterisk-users] Agi script for working hours PBX   mahesh katta
• [asterisk-users] Festival VOICE MAIL TO FEMAIL   mahesh katta
• [asterisk-users] benefits of asterisk 1.8   vip killa
• [asterisk-users] benefits of asterisk 1.8   vip killa
• [asterisk-users] asterisk 1.6 - 511 Command not permitted causing high CPU usage   vip killa
• [asterisk-users] benefits of asterisk 1.8   vip killa
• [asterisk-users] change destination on digit   vip killa
• [asterisk-users] fast AGI memory leaks   vip killa
• [asterisk-users] asterisk voicemail distribution groups   vip killa
• [asterisk-users] asterisk voicemail distribution groups   vip killa
• [asterisk-users] DTMF begin ignored   vip killa
• [asterisk-users] call paging interrupts call when using Mitel 5224   vip killa
• [asterisk-users] call paging interrupts call when using Mitel 5224   vip killa
• [asterisk-users] call paging interrupts call when using Mitel 5224   vip killa
• [asterisk-users] call paging interrupts call when using Mitel 5224   vip killa
• [asterisk-users] call paging interrupts call when using Mitel 5224   vip killa
• [asterisk-users] call paging interrupts call when using Mitel 5224   vip killa
• [asterisk-users] call paging interrupts call when using Mitel 5224   vip killa
• [asterisk-users] call paging interrupts call when using Mitel 5224   vip killa
• [asterisk-users] benefits of asterisk 1.8   Gopal krishnan
• [asterisk-users] Asterisk and Audiocodes PRI card   Gopal krishnan
• [asterisk-users] RTP Streaming   Gopal krishnan
• [asterisk-users] Integration of OpenVXI   Gopal krishnan
• [asterisk-users] SIP Register DOS attack   Al lists
• [asterisk-users] Access Voicemail Asterisk 1.8 FreeBSD 8.2   motty.cruz
• [asterisk-users] Asterisk 1.6 Dahdi on Centos 5.2   motty.cruz
• [asterisk-users] Asterisk 1.6 Dahdi on Centos 5.2   motty.cruz
• [asterisk-users] Asterisk 1.6 Dahdi on Centos 5.2   motty.cruz
• [asterisk-users] skinny and 7961   Kelly opal
• [asterisk-users] Asterisk 1.8.4.2 Now Available (Security Release)   satish patel
• [asterisk-users] asterisk logger permission   satish patel
• [asterisk-users] Queue base polycom custom ringtype   satish patel
• [asterisk-users] benefits of asterisk 1.8   satish patel
• [asterisk-users] benefits of asterisk 1.8   satish patel
• [asterisk-users] benefits of asterisk 1.8   satish patel
• [asterisk-users] broken SVN asterisk 1.8 ?   satish patel
• [asterisk-users] asterisk 1.8 issue with polycom dialplan   satish patel
• [asterisk-users] asterisk 1.8 issue with polycom dialplan   satish patel
• [asterisk-users] PRI issue its BUSY   satish patel
• [asterisk-users] PRI issue its BUSY   satish patel
• [asterisk-users] PRI issue its BUSY   satish patel
• [asterisk-users] PRI issue its BUSY   satish patel
• [asterisk-users] PRI issue its BUSY   satish patel
• [asterisk-users] [SOLVED]PRI issue its BUSY   satish patel
• [asterisk-users] reload chan_dahdi.conf without disconnect active calls   satish patel
• [asterisk-users] reload chan_dahdi.conf without disconnect active calls   satish patel
• [asterisk-users] PRI hangup request, cause 18   satish patel
• [asterisk-users] PRI hangup request, cause 18   satish patel
• [asterisk-users] Asterisk 1.8 broken MWI   satish patel
• [asterisk-users] Asterisk 1.8 broken MWI   satish patel
• [asterisk-users] Asterisk 1.8 broken MWI   satish patel
• [asterisk-users] Asterisk 1.8 broken MWI   satish patel
• [asterisk-users] Asterisk 1.8 broken MWI   satish patel
• [asterisk-users] Asterisk 1.8 broken MWI   satish patel
• [asterisk-users] Asterisk 1.8 broken MWI   satish patel
• [asterisk-users] Asterisk 1.8 broken MWI   satish patel
• [asterisk-users] Interesting PRI issue   satish patel
• [asterisk-users] issues.asterisk.org/jira not working   satish patel
• [asterisk-users] How asterisk use pri channel   satish patel
• [asterisk-users] Asterisk 1.8 broken MWI   satish patel
• [asterisk-users] Asterisk 1.8 broken MWI   satish patel
• [asterisk-users] Asterisk 1.8 broken MWI   satish patel
• [asterisk-users] Polycom 501 Settings/subscription expiry   satish patel
• [asterisk-users] asterisk 1.8 PRI random call drop issue   satish patel
• [asterisk-users] asterisk 1.8 PRI random call drop issue   satish patel
• [asterisk-users] Asterisk call limitation   satish patel
• [asterisk-users] SMS with Asterisk   jon pounder
• [asterisk-users] asterisk fails when DNS or internet fails   nhadie ramos
• [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel   randall
• [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel   randall
• [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel   randall
• [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel   randall
• [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel   randall
• [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel   randall
• [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel   randall
• [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel   randall
• [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel   randall
• [asterisk-users] Voxbone numbers   randulo
• [asterisk-users] Friday: Sipgate, Yate and Astricon   randulo
• [asterisk-users] Sending SMS on Friday at 12 Noon EDT   randulo
• [asterisk-users] Sending SMS on Friday at 12 Noon EDT   randulo
• [asterisk-users] SMS with Asterisk   randulo
• [asterisk-users] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko   randulo
• [asterisk-users] Inbound SMS   randulo
• [asterisk-users] Google Voice receiving call problem   randulo
• [asterisk-users] eComm - Are you going to San Francisco next week?   randulo
• [asterisk-users] Google Voice receiving call problem   randulo
• [asterisk-users] OT - Polycom - Which provisioning protocol to choose ?   randulo
• [asterisk-users] connecting to SIP Provider with virtual IP from pacemaker cluster   rosenberger at taoweb.at
• [asterisk-users] Can I use phone line to recive faxes?   khalid touati
• [asterisk-users] SIP Register DOS attack   khalid touati
• [asterisk-users] Can I use phone line to recive faxes?   khalid touati
• [asterisk-users] Problem with ReceiveFAX app from FFA   khalid touati
• [asterisk-users] Problem with ReceiveFAX app from FFA   khalid touati
• [asterisk-users] Problem with ReceiveFAX app from FFA   khalid touati
• [asterisk-users] Problem with ReceiveFAX app from FFA   khalid touati
• [asterisk-users] menu issue   khalid touati
• [asterisk-users] To know if the ISDN PRI E1 is UP?   James zhu
• [asterisk-users] reload chan_dahdi.conf without disconnect active calls   James zhu
• [asterisk-users] PRI issue its BUSY   James zhu
• [asterisk-users] Can I use phone line to recive faxes?   James zhu
• [asterisk-users] Interesting PRI issue   James zhu

Last message date: Thu Jun 30 19:42:13 CDT 2011
Archived on: Thu Jun 30 19:45:27 CDT 2011

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