Hello,
'sip show channel' also does not give this info.
sip show channel f600ed29f561d57
localhost*CLI
* SIP CallI
Curr. trans. direction: Incoming
Call-ID:f600ed29f561d57f
Owner channel ID: SIP/100-
Our Codec Capability: 14
Non-Codec Capability
Hi all
I've been saddled with recreating a running Asterisk PBX setup (with Ruby on
Rails). Due to some wrangling between my client and the original developers I
am not able to talk to the developers themselves but have been given full SSH
access to their servers!
My questions are regarding
Hi,
Not sure why you didnt get it, when I did thta command for originator
channel it showed me the CDR variables list which included
CDR Variables:
level 1: dnid=
level 1: clid=XXX
level 1: src=
level 1: dst=
level 1: dcontext=SIP-incoming
level 1: channel=
level 1:
On Tue, 2011-12-13 at 16:32 -0800, Edwin Lam wrote:
On 12/10/11 9:54 PM, Takehiro Matsushima wrote:
I'd configured realtime registration, but configuration was not applied
when I
changed a row of sippeers table.
To apply, 'sip reload' was needed (in Asterisk 1.8.0).
or you can 'sip
In article CAJUJwthT=mpyxq+omt5hrextl1iqvd0kbs+jhtqlvsqscay...@mail.gmail.com,
Sammy Govind govoi...@gmail.com wrote:
Hi,
Not sure why you didnt get it, when I did thta command for originator
channel it showed me the CDR variables list which included
That's from show channel, not sip show
oops, you got it.
On Wed, Dec 14, 2011 at 2:43 PM, Tony Mountifield t...@softins.co.ukwrote:
In article CAJUJwthT=
mpyxq+omt5hrextl1iqvd0kbs+jhtqlvsqscay...@mail.gmail.com,
Sammy Govind govoi...@gmail.com wrote:
Hi,
Not sure why you didnt get it, when I did thta command for originator
On Thu, 1 Dec 2011 14:09:29 +0300, James Mutuku listmut...@gmail.com
wrote:
I have worked with bare asterisk + freepbx before. the mypbx was just
an example but my reference to appliances as a whole.
The appliances seem to have lower entry costs.
Appliances have less RAM + storage, so you'll
finally I got it with 'core show channel' channel-id
thanks for your support.
Date: Wed, 14 Dec 2011 15:11:49 +0500
From: govoi...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] get start-time of all active calls
oops, you got it.
On Wed, Dec 14, 2011 at
On 14-12-11 10:18, Brynjolfur Thorvardsson wrote:
Hi all
I’ve been saddled with recreating a running Asterisk PBX setup (with
Ruby on Rails). Due to some wrangling between my client and the original
developers I am not able to talk to the developers themselves but have
been given full SSH
Hello list,
An Asterisk installation that was doing fine suddenly stared segfaulting
a couple of times per day. I enabled all the logging and debugging to
try to find a pattern but there was too much information to see exactly
where it broke. So I enabled core dump and did backtraces and all
On 14 December 2011 12:56, Paulo Santos paulo.r.san...@sapo.pt wrote:
Hello list,
An Asterisk installation that was doing fine suddenly stared segfaulting a
couple of times per day. I enabled all the logging and debugging to try to
find a pattern but there was too much information to see
On 14-12-11 13:56, Paulo Santos wrote:
Hello list,
An Asterisk installation that was doing fine suddenly stared segfaulting
a couple of times per day. I enabled all the logging and debugging to
try to find a pattern but there was too much information to see exactly
where it broke. So I enabled
Hi, thanks for your answer. I suppose that both the STUN servers and ActiveMQ
are there to give a better/more reliable service which is obviously a good idea.
From trying to find out some more on the Internet I get the idea that CSTele
might have something to do with Circuit Switching. I am
Hello,
Thank you all for the replies.
Steve Davies wrote:
If I was guessing, I'd say that the channel structure that is being
modified by the ast_setstate() call is incomplete, and contains some
garbage pointers.
If I was guessing further, I'd say that the callerID pointers are
the most
On Wed, Dec 14, 2011 at 2:18 AM, Brynjolfur Thorvardsson bi...@itanet.nuwrote:
**
I’ve been saddled with recreating a running Asterisk PBX setup (with Ruby
on Rails). Due to some wrangling between my client and the original
developers I am not able to talk to the developers themselves but
Any thoughts on what could be causing this ?
--
Thanks, Phil
- Original Message -
Okay, though removing the space and reloading the module still throws
the same error messages.
--
Thanks, Phil
- Original Message -
Generally speaking, no. if you need the space, use
Hi Carlos and thanks for your answer. To begin with: I am a noob in all
telephony/asterisk/ror fields, coming from a Classic ASP/MS background! I've
been nosing around in RoR and Asterisk for the last month or so and have
managed to create several RoR sites and to get an Asterisk server up and
Getting involved in an existing, and possibly broken system is the wrong
way to start with Asterisk. I know, because that's how my career in VoIP
started. I had to unlearn a lot of poor practices I learned from that
system.
But anyway without prior documentation or the ability to get the
Hi Carlos and thanks for the advice. I agree with you wholeheartedly but I'm
not sure if I have much choice in the matter. The system was originally
designed to offer PBX services to private clinics and currently handles between
10 and 20, with 70 phone numbers. The guys I work for want to
Please feel free to pass this along:
DON'T DO IT!
Taking questionable code, from what appears to be a questionable
relationship, and then trying to extend its life is probably the craziest
way to go about this.
You, personally, are in for a steep learning curve on this. Having worked
with
You are 110% correct Carlos, but Im sure B.T. likes to eat. We all have to
do things we dont like.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Wednesday, December 14, 2011 10:29 AM
To: Asterisk Users
21 matches
Mail list logo