Re: [asterisk-users] Hangup Cause and SIP Response Code

2012-04-25 Thread Eric Wieling
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Wednesday, April 25, 2012 6:25 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hangup Cause and SIP Response Code On

Re: [asterisk-users] Hangup Cause and SIP Response Code

2012-04-25 Thread Kevin P. Fleming
On 04/25/2012 04:45 PM, brya...@zktech.com wrote: Kevin I am using 1.8.x& 10.x Then you have SIP_CAUSE available, although you'll have to enable it because it is off by default due to performance concerns. Bryant Zimmerman (ZK Tech Inc./interNetGR) (616) 855-1030 Ext. 2003 On Apr 25, 2

[asterisk-users] Restart single dahdi span

2012-04-25 Thread James Lamanna
Hi, Is it possible yet to restart a single Dahdi span in any version of Asterisk? (instead of all of them) Thanks. -- James -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for

Re: [asterisk-users] Hangup Cause and SIP Response Code

2012-04-25 Thread BryantZ
Kevin I am using 1.8.x & 10.x Bryant Zimmerman (ZK Tech Inc./interNetGR) (616) 855-1030 Ext. 2003 On Apr 25, 2012, at 5:00 PM, "Kevin P. Fleming" wrote: > On 04/25/2012 07:08 AM, Bryant Zimmerman wrote: >> I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to >> track the actu

Re: [asterisk-users] Hangup Cause and SIP Response Code

2012-04-25 Thread Kevin P. Fleming
On 04/25/2012 07:08 AM, Bryant Zimmerman wrote: I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to track the actual SIP response code as well. How do I get access to it durring the hangup? It's rather hard to answer that question without at least knowing what version of Aste

[asterisk-users] Open source replacement for AudioCodes nCite 1000 SBC

2012-04-25 Thread Matthew J. Roth
List users, I have an AudioCodes nCite 1000 SBC that is end-of-life and I'm looking to replace it with open source software. I believe one of the SIP proxy projects will fit my needs, but I'm a bit overwhelmed by the number of choices and I'd like the advice of experienced users before I venture

Re: [asterisk-users] meetme identify user number

2012-04-25 Thread Dan Austin
Daniel wrote: > Hi Group, > is in MeetMe any option to identify the own number (from the view of a > caller)? > I would like to write an option to set on the telephone an request for voice, > if the room > is muted. That request should display on our Conference Control > Website and an Admin

Re: [asterisk-users] CONNECTEDLINE() updated during SIP events?

2012-04-25 Thread Kevin P. Fleming
On 04/25/2012 11:54 AM, Steve Davies wrote: A further question... It appears that for SIP endpoints, this facility only updates RPID and PAI headers? I have found that there appear to be 4 different SIP CID-update mechanisms "out there" as follows: - Update RPID and PAI (ITSP and trunks often u

Re: [asterisk-users] CONNECTEDLINE() updated during SIP events?

2012-04-25 Thread Steve Davies
On 25 April 2012 16:55, Richard Mudgett wrote: [snip] > >> - Is it possible to have the COLP/COLR information updated when a SIP >> attended transfer is completed? If so how? > > Transfers generate connected line update events automatically.  The connected > line interception macros give you a cha

Re: [asterisk-users] CONNECTEDLINE() updated during SIP events?

2012-04-25 Thread Richard Mudgett
> I have read the excellent information here: > > https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information > and believe I have an understanding of what is offered. I have a > couple of questions: > > - Is it possible to update COLP/COLR when a SIP redirect occurs, or > wh

[asterisk-users] Asterisk - Nortel transfer problem

2012-04-25 Thread Mc GRATH Ricardo
Hi Carlos It could help if you can get a trace of the call transfer from Nortel to SIP extension on the Asterisk (1303), if no way to get from Nortel get from Asterisk. I guest operator try to make a bind call transfer, without wait complete DR2 signalling exchange at least minimal time exchang

[asterisk-users] CONNECTEDLINE() updated during SIP events?

2012-04-25 Thread Steve Davies
Hi, I have read the excellent information here: https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information and believe I have an understanding of what is offered. I have a couple of questions: - Is it possible to update COLP/COLR when a SIP redirect occurs, or when a SIP div

[asterisk-users] Hangup Cause and SIP Response Code

2012-04-25 Thread Bryant Zimmerman
I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to track the actual SIP response code as well. How do I get access to it durring the hangup? Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http:/

Re: [asterisk-users] Strange problem on ougoing call

2012-04-25 Thread Olivier CALVANO
Ok thanks i test. I put "match_auth_username=yes" on the two server ? And for insecure, into the realtime database ? or into sip.conf of the second server ? best regards olivier Le 25 avril 2012 09:34, Leandro Dardini a écrit : > > > 2012/4/25 Olivier CALVANO >> >> Sure, sorry for the Confu

[asterisk-users] chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 returned -1: Operation not permitted ??

2012-04-25 Thread Olivier CALVANO
Hi i have a lot of error in the CLI of one of my Asterisk: [Apr 25 09:30:46] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation not permitted [Apr 25 09:30:46] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8639de8 (

Re: [asterisk-users] Strange problem on ougoing call

2012-04-25 Thread Leandro Dardini
2012/4/25 Olivier CALVANO > Sure, sorry for the Confusion ;=) > > > > > Server A "Trader": > Asterisk Server 1.6.x for call routing only. > IP Adress: 172.16.0.11 > Use Realtim on MySQL Database > This server route all call to a lot of VoIP Carrier. > > > Server B "Ipbx" >

Re: [asterisk-users] Strange problem on ougoing call

2012-04-25 Thread Olivier CALVANO
Sure, sorry for the Confusion ;=) Server A "Trader": Asterisk Server 1.6.x for call routing only. IP Adress: 172.16.0.11 Use Realtim on MySQL Database This server route all call to a lot of VoIP Carrier. Server B "Ipbx" Asterisk Server 1.6.x for connect a lo