-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
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
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
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
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
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
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
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
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
> 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
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
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
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:/
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
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 (
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"
>
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
17 matches
Mail list logo