Re: [asterisk-users] multiple users for jabber.conf

2012-09-11 Thread Hans Witvliet
On Wed, 2012-09-12 at 00:01 -0500, Vladimir Mikhelson wrote: > Hans, > > I did not try 10 or 11 as I run 1.8.15. Following are the related > conf files. > > gtalk.conf > > [General] > context = default > allowguest = yes ; Required if you want to accept calls > from people Not on yo

[asterisk-users] [asterisk-user] INTERNAL_OBJ error in asterisk 1.8.13

2012-09-11 Thread Chandrakant Solanki
Hi All, Asterisk Version: 1.8.13.0 CentOs : 6.3 Continues getting this error while submitting cdr record. [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb

Re: [asterisk-users] Asterisk HangUp not breaking incoming call for caller

2012-09-11 Thread Vladimir Mikhelson
On 9/12/2012 12:05 AM, Raj Mathur (??? ?) wrote: > On Wednesday 12 Sep 2012, Vladimir Mikhelson wrote: >> Raj, >> >> I am just confirming it happens here as well. >> >> CentOS 5.7. Asterisk 1.8.15.1. DAHDI 2.6.1. >> >> Digium, Inc. Wildcard TDM410 4-port analog card (rev 11) >> >> Loadzone = u

Re: [asterisk-users] Asterisk HangUp not breaking incoming call for caller

2012-09-11 Thread Mitul Limbani
Raj, Problem 1 is where asterisk times out and line gets free However for problem 2 There is no way that this line frees up, as it depends upon remote side infra of caller, if they calling from pri ckt they possibly could identify our hangup signal, but if they calling from Analog exchange this

Re: [asterisk-users] Asterisk HangUp not breaking incoming call for caller

2012-09-11 Thread Raj Mathur (राज माथुर)
On Wednesday 12 Sep 2012, Vladimir Mikhelson wrote: > Raj, > > I am just confirming it happens here as well. > > CentOS 5.7. Asterisk 1.8.15.1. DAHDI 2.6.1. > > Digium, Inc. Wildcard TDM410 4-port analog card (rev 11) > > Loadzone = us > > The problem started manifesting itself after I switche

Re: [asterisk-users] multiple users for jabber.conf

2012-09-11 Thread Vladimir Mikhelson
Hans, I did not try 10 or 11 as I run 1.8.15. Following are the related conf files. *gtalk.conf* [General] context = default allowguest = yes ; Required if you want to accept calls from people Not on your contact list. bindaddr= ;; These two settings are very critical for getting

Re: [asterisk-users] Asterisk HangUp not breaking incoming call for caller

2012-09-11 Thread Mitul Limbani
This has been happening since the asterisk 1.2 days, makes me believe it has something to do with Analog FXO ckts provided. Mitul Limbani On Sep 12, 2012 10:18 AM, "Vladimir Mikhelson" wrote: > Raj, > > I am just confirming it happens here as well. > > CentOS 5.7. Asterisk 1.8.15.1. DAHDI 2.6.1.

Re: [asterisk-users] Asterisk HangUp not breaking incoming call for caller

2012-09-11 Thread Vladimir Mikhelson
Raj, I am just confirming it happens here as well. CentOS 5.7. Asterisk 1.8.15.1. DAHDI 2.6.1. Digium, Inc. Wildcard TDM410 4-port analog card (rev 11) Loadzone = us The problem started manifesting itself after I switched to 1.8.x from 1.6.2.x Typical scenario: a caller apparently hangs up, d

Re: [asterisk-users] asterisk boxes looses registration

2012-09-11 Thread Eric Wieling
Try adding qualify=yes -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Tuesday, September 11, 2012 3:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users]

Re: [asterisk-users] html/js/flash/air SIP clients?

2012-09-11 Thread Matt Riddell
On 7/08/2012, at 7:38 PM, Arstan Jusupov wrote: > Correct me if I'm wrong but phono works with voxeo tropo. Phono can also be used to make SIP calls directly to and from your Asterisk servers via Voxeo. I use it in my CRM package to provide a softphone that logs into call queues and makes/rec

[asterisk-users] multiple users for jabber.conf

2012-09-11 Thread Hans Witvliet
Hi all, Been reading about chan_motif / chan_xmpp in the wiki's for 1.8, 10 and 11 version of asterisk. In each example i got the impression that the asterisk server is registering on a XMPP server as a single user with the credentials as specified in jabber.conf. Instead of a single xmpp-user, c

[asterisk-users] asterisk boxes looses registration

2012-09-11 Thread Jerry Geis
I have a couple asterisk boxes, running sip between both boxes. 1.4.43 on both. both are installed from source, both have default settings, My config for one box is: [devgeis] type=friend defaultname=devgeis username=devgeis secret=yes disallow=all allow=ulaw allow=alaw allow=gsm rtptimeout=60

[asterisk-users] Linebreaks in cdr_custom.conf / cdr_sqlite3_custom.conf

2012-09-11 Thread Stefan at WPF
In the cdr_custom.conf / cdr_sqlite3_custom.conf configuration files, is it somehow possible to split the enumeration of CDR fields over multiple lines? I get parsing errors when using more then a single line, but a single line is very confusing if one has many CDR fields. So, is there something li

[asterisk-users] Asterisk HangUp not breaking incoming call for caller

2012-09-11 Thread Raj Mathur (राज माथुर)
Hi, Asterisk 1.8.13 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express) When Asterisk executes HangUp() on an incoming call, the line remains connected for the caller. Zone = in, opermode = INDIA. Line set to fxsks. Any help on where to start

[asterisk-users] Asterisk HangUp not breaking incoming call for caller

2012-09-11 Thread Raj Mathur (राज माथुर)
Hi, Asterisk 1.8.13 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express) When Asterisk executes HangUp() on an incoming call, the line remains connected for the caller. Zone = in, opermode = INDIA. Line set to fxsks. Any help on where to start

[asterisk-users] codec priorities

2012-09-11 Thread Jeff LaCoursiere
Hello, I am about to start playing with wideband codecs in our lab, and was hoping to get some clarification on a few things. To date I've pretty much forced the use of G.711 on all legs of all calls, and life has been grand. Now we are distributing phones with G.722 and speex capability, and I

Re: [asterisk-users] Async AGI

2012-09-11 Thread Pavel Siderov
Hi David, For sure I will ask them but I think Asterisk should be able to handle this case because it doesn't matter if it is adhearsion or something else. If it is not present there is no way to get answer. So there must be some way to go to another priority in the dialplan. @Danny - this is the