Re: [asterisk-users] asterisk-users Digest, Vol 73, Issue 58
On 8/27/10, asterisk-users-requ...@lists.digium.com wrote: > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > asterisk-users-requ...@lists.digium.com > > You can reach the person managing the list at > asterisk-users-ow...@lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > >1. CDR on Transfer... (Carlos Chavez) >2. Re: Asterisk 1.6.1.17 ACK/BYE question (Trevor Benson) >3. Re: Use of AGISIGHUP (Danny Nicholas) >4. double DTMF digits (M S) >5. Re: double DTMF digits (Andres) >6. Re: Use of AGISIGHUP (Steve Edwards) >7. Re: Use of AGISIGHUP (Danny Nicholas) >8. Re: Use of AGISIGHUP (Steve Edwards) >9. Re: double DTMF digits (Matt Desbiens) > 10. Asterisk 1.6 Displaying in Debug that it is playing a ulaw > file using BackGround() but no audio is heard from the phone > (Joe Wood) > 11. Re: double DTMF digits (M S) > 12. Re: Use of AGISIGHUP (Lee Archer) > 13. dynamic MeetMe, min. digits (Xavier) > 14. Re: dynamic MeetMe, min. digits (Doug Lytle) > 15. Re: dynamic MeetMe, min. digits (Xavier D.) > 16. music on hold in blind transfer (Tino) > 17. queue agent and blind transfer (Tino) > 18. Call Forwarding (Dan Journo) > 19. Re: music on hold in blind transfer (Paul Belanger) > 20. Re: Call Forwarding (Stefan Schmidt) > 21. Duplicate channel variables after transfer (Alex Hermann) > 22. Re: CDR on Transfer... (Andra?) > > > -- > > Message: 1 > Date: Thu, 26 Aug 2010 12:25:07 -0500 > From: Carlos Chavez > Subject: [asterisk-users] CDR on Transfer... > To: Asterisk > Message-ID: <1282843507.2830.13.ca...@cursor.telecomabmex.com> > Content-Type: text/plain; charset="utf-8" > > I have searched for some time but I have not found an asnwer on how to > fix the CDR when a call is transferred. The problem is that if someone > dials a cell phone and then transfers the call to another extensi?n the > CDR for the cell call stops and there is no way to track that the call > was transferred so we can bill correctly. Many people have asked this > question but there is no answer, only a mention that it should be fixed > in 1.6 which it is not (at least on 1.6.2.11). > > Any tips oh how to correct this problem? A lot of customers give me > grief about this because they cannot properly bill people for their cell > calls. > > -- > Telecomunicaciones Abiertas de M?xico S.A. de C.V. > Carlos Ch?vez Prats > Director de Tecnolog?a > +52-55-91169161 ext 2001 > -- next part -- > A non-text attachment was scrubbed... > Name: not available > Type: application/pgp-signature > Size: 198 bytes > Desc: This is a digitally signed message part > Url : > http://lists.digium.com/pipermail/asterisk-users/attachments/20100826/f970d9ee/attachment-0001.pgp > > -- > > Message: 2 > Date: Thu, 26 Aug 2010 10:30:16 -0700 > From: Trevor Benson > Subject: Re: [asterisk-users] Asterisk 1.6.1.17 ACK/BYE question > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <65f20266-3e46-4dcc-a17d-d181f8e4a...@a-1networks.com> > Content-Type: text/plain; charset="windows-1252" > > We have a box running 1.6.2.11 on CentOS 5 using the RPM's from the Digium > CentOS repository. We just left a 60 second voicemail on the system and had > the full audio as well in the inbox. Not sure how your SIP configuration > ties your SBC in, but native "users" created via users.conf and sip.conf > appears to be working for me. Wouldnt be able to test more without knowing > what settings you had between Asterisk and the SBC. > > > -- > Trevor Benson > dCAP, LPIC-1, CLA, Network+, MCP, CNA > A1 Networks - Network Engineer > DID (707)703-1041 > FAX (707)703-1983 > > > > > > > On Aug 26, 2010, at 8:47 AM, Steven C. Blair wrote: > >> >> As a test we built Asterisk v1.6.2.11 on a new server. This version of >> Asterisk exhibits the same behavior. From ngrep?s perspective we see an >> ACK followed immediately by a BYE message. The user hears the recording >> being played, begins to leave a message and is disconnected about 10 >> seconds into the call. >> >> >> >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven C. >> Blair >> Sent: Wednesday, August 25, 2010 2:08 PM >> To: asterisk-users@lists.digium.com >> Subject: [asterisk-users] Asterisk 1.6.1.17 ACK/BYE question >> >> >> We?re running Asterisk 1.6.1.17 for our campus voicemail server and >> Juniper M120s as our SBC. Unanswered calls, which arrive via the SBC, are >> diverted to
Re: [asterisk-users] asterisk-users Digest, Vol 73, Issue 58
On 8/27/10, asterisk-users-requ...@lists.digium.com wrote: > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > asterisk-users-requ...@lists.digium.com > > You can reach the person managing the list at > asterisk-users-ow...@lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > >1. CDR on Transfer... (Carlos Chavez) >2. Re: Asterisk 1.6.1.17 ACK/BYE question (Trevor Benson) >3. Re: Use of AGISIGHUP (Danny Nicholas) >4. double DTMF digits (M S) >5. Re: double DTMF digits (Andres) >6. Re: Use of AGISIGHUP (Steve Edwards) >7. Re: Use of AGISIGHUP (Danny Nicholas) >8. Re: Use of AGISIGHUP (Steve Edwards) >9. Re: double DTMF digits (Matt Desbiens) > 10. Asterisk 1.6 Displaying in Debug that it is playing a ulaw > file using BackGround() but no audio is heard from the phone > (Joe Wood) > 11. Re: double DTMF digits (M S) > 12. Re: Use of AGISIGHUP (Lee Archer) > 13. dynamic MeetMe, min. digits (Xavier) > 14. Re: dynamic MeetMe, min. digits (Doug Lytle) > 15. Re: dynamic MeetMe, min. digits (Xavier D.) > 16. music on hold in blind transfer (Tino) > 17. queue agent and blind transfer (Tino) > 18. Call Forwarding (Dan Journo) > 19. Re: music on hold in blind transfer (Paul Belanger) > 20. Re: Call Forwarding (Stefan Schmidt) > 21. Duplicate channel variables after transfer (Alex Hermann) > 22. Re: CDR on Transfer... (Andra?) > > > -- > > Message: 1 > Date: Thu, 26 Aug 2010 12:25:07 -0500 > From: Carlos Chavez > Subject: [asterisk-users] CDR on Transfer... > To: Asterisk > Message-ID: <1282843507.2830.13.ca...@cursor.telecomabmex.com> > Content-Type: text/plain; charset="utf-8" > > I have searched for some time but I have not found an asnwer on how to > fix the CDR when a call is transferred. The problem is that if someone > dials a cell phone and then transfers the call to another extensi?n the > CDR for the cell call stops and there is no way to track that the call > was transferred so we can bill correctly. Many people have asked this > question but there is no answer, only a mention that it should be fixed > in 1.6 which it is not (at least on 1.6.2.11). > > Any tips oh how to correct this problem? A lot of customers give me > grief about this because they cannot properly bill people for their cell > calls. > > -- > Telecomunicaciones Abiertas de M?xico S.A. de C.V. > Carlos Ch?vez Prats > Director de Tecnolog?a > +52-55-91169161 ext 2001 > -- next part -- > A non-text attachment was scrubbed... > Name: not available > Type: application/pgp-signature > Size: 198 bytes > Desc: This is a digitally signed message part > Url : > http://lists.digium.com/pipermail/asterisk-users/attachments/20100826/f970d9ee/attachment-0001.pgp > > -- > > Message: 2 > Date: Thu, 26 Aug 2010 10:30:16 -0700 > From: Trevor Benson > Subject: Re: [asterisk-users] Asterisk 1.6.1.17 ACK/BYE question > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <65f20266-3e46-4dcc-a17d-d181f8e4a...@a-1networks.com> > Content-Type: text/plain; charset="windows-1252" > > We have a box running 1.6.2.11 on CentOS 5 using the RPM's from the Digium > CentOS repository. We just left a 60 second voicemail on the system and had > the full audio as well in the inbox. Not sure how your SIP configuration > ties your SBC in, but native "users" created via users.conf and sip.conf > appears to be working for me. Wouldnt be able to test more without knowing > what settings you had between Asterisk and the SBC. > > > -- > Trevor Benson > dCAP, LPIC-1, CLA, Network+, MCP, CNA > A1 Networks - Network Engineer > DID (707)703-1041 > FAX (707)703-1983 > > > > > > > On Aug 26, 2010, at 8:47 AM, Steven C. Blair wrote: > >> >> As a test we built Asterisk v1.6.2.11 on a new server. This version of >> Asterisk exhibits the same behavior. From ngrep?s perspective we see an >> ACK followed immediately by a BYE message. The user hears the recording >> being played, begins to leave a message and is disconnected about 10 >> seconds into the call. >> >> >> >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven C. >> Blair >> Sent: Wednesday, August 25, 2010 2:08 PM >> To: asterisk-users@lists.digium.com >> Subject: [asterisk-users] Asterisk 1.6.1.17 ACK/BYE question >> >> >> We?re running Asterisk 1.6.1.17 for our campus voicemail server and >> Juniper M120s as our SBC. Unanswered calls, which arrive via the SBC, are >> diverted to
