Re: [asterisk-users] pjsip: Inbound calls: selecting the correct trunk with one provider and different numbers

On Fri, Jun 9, 2017, at 03:44 PM, Michael Maier wrote: > > Further investigation showed, that Telekom provides the line info in the > Request Line (as seen by Wireshark): > > Request-Line: INVITE sip:+49@46.37.15.4:5060;line=azpreyb SIP/2.0 > > You can't find it if you expect it in

Re: [asterisk-users] Working around missing libmyodbc in Debian Stretch

I found this [1], downloaded a tar file from [2], hand copied a single libmyodbc5a.so into appropriate /usr/lib/x86_64-linux-gnu/odbc/ directory and adapted /etc/odbcinst.ini file accordingly. After all this, it seemed to work OK but I'm still not too confident. Shall I trust a couple of

[asterisk-users] Asterisk 13 attended transfer alcatel

Hello, Since upgrading from asterisk 11 to asterisk 13 (I have tested up to the latest 13.16.0 release), we have a problem with attended transfers to an alcatel pbx in which the call being transferred still has music on hold even after the transfer has completed. Is this a known issue? Any new

Re: [asterisk-users] Asterisk 13 attended transfer alcatel

On Fri, Jun 9, 2017, at 04:59 AM, Jason TOMLINSON wrote: > Hello, > > Since upgrading from asterisk 11 to asterisk 13 (I have tested up to the > latest 13.16.0 release), we have a problem with attended transfers to an > alcatel pbx in which the call being transferred still has music on hold >

Re: [asterisk-users] pjsip user_eq_phone adds user=phone to anonymous user bug?

On Fri, Jun 09, 2017 at 11:40:01AM -0300, Joshua Colp wrote: > What seems to be happening is that the session is being set up and the > user=phone parameter added. It's only after that the values are updated > to be Anonymous and the user=phone parameter is left there. Please file > an issue[1]

[asterisk-users] pjsip user_eq_phone adds user=phone to anonymous user bug?

With pjsip (asterisk 13.14.1) I see the problem that an anonymous from header gets user=phone appendend to the URI if user_eq_phone=yes is specified: On the incoming leg: From: anonymous ;tag=Q5zBj7BMnvI6Fe6O2866fox3ZHmn-smt Get transformed to From:

Re: [asterisk-users] pjsip user_eq_phone adds user=phone to anonymous user bug?

On Fri, Jun 9, 2017, at 11:30 AM, Daniel Tryba wrote: > With pjsip (asterisk 13.14.1) I see the problem that an anonymous from > header gets user=phone appendend to the URI if user_eq_phone=yes is > specified: > > On the incoming leg: > From: anonymous >

Re: [asterisk-users] Asterisk 13 attended transfer alcatel

How are both machines connected to each other ? Through a SIP trunk ? A TDM one ? 2017-06-09 9:59 GMT+02:00 Jason TOMLINSON : > Hello, > > > > Since upgrading from asterisk 11 to asterisk 13 (I have tested up to the > latest 13.16.0 release), we have a problem with

Re: [asterisk-users] Upgraded server crashes on voicemail storage

Well, I guess my assumption has been proven wrong. It is NOT the odbc drive. I recompiled Asterisk w/o odbc voicemail storage and I'm still getting crashes when someone leave voicemail. I tried to run strace on the server, but didn't get much: = voip11 ~ #

Re: [asterisk-users] pjsip: Inbound calls: selecting the correct trunk with one provider and different numbers

On 06/09/2017 at 08:44 PM Michael Maier wrote: > On 06/08/2017 at 10:22 PM Michael Maier wrote: >> Hello Joshua, >> >> thank you very much for your extremely quick answer! I really appreciate >> your work and your friendly and your patient support! >> >> >> On 06/07/2017 at 10:33 PM, Joshua Colp

Re: [asterisk-users] pjsip: Inbound calls: selecting the correct trunk with one provider and different numbers

On 06/08/2017 at 10:22 PM Michael Maier wrote: > Hello Joshua, > > thank you very much for your extremely quick answer! I really appreciate > your work and your friendly and your patient support! > > > On 06/07/2017 at 10:33 PM, Joshua Colp wrote: >> On Wed, Jun 7, 2017, at 05:28 PM, Michael

Re: [asterisk-users] Upgraded server crashes on voicemail storage

On 06/09/2017 12:37 PM, Mike Diehl wrote: Well, I guess my assumption has been proven wrong. It is NOT the odbc drive. I recompiled Asterisk w/o odbc voicemail storage and I'm still getting crashes when someone leave voicemail. This is probably not it BUT. A long time ago, voicemail lost