similar to: One way audio on chan_gtalk

Displaying 20 results from an estimated 1000 matches similar to: "One way audio on chan_gtalk"

2010 Nov 30
2
Error loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object file: No such file or directory
Hello, Can't get chan_gtalk.so module to load, neither res_jabber.so: Asterisk*CLI> module load chan_gtalk.so Unable to load module chan_gtalk.so Command 'module load chan_gtalk.so ' failed. [Dec 1 16:10:05] WARNING[2931]: loader.c:387 load_dynamic_module: Error loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object file: No such file or directory [Dec
2011 Jul 18
1
chan_gtalk load error
Hi, When starting Asterisk (1.8.5.0) I see in messages: [Jul 18 15:47:50] WARNING[15491] loader.c: Error loading module 'chan_gtalk.so': /usr/lib/asterisk/modules/chan_gtalk.so: undefined symbol: ast_aji_get_client [Jul 18 15:47:50] WARNING[15491] loader.c: Module 'chan_gtalk.so' could not be loaded. Yet I do have iksemel installed: ls -l /usr/local/lib/libik* -rw-r--r-- 1
2011 Nov 28
1
centos asterisk 1.8 rpms: chan_gtalk and res_jabber missing?
Hi All, While I'm certainly comfortable compiling from sources, I'm trying to do an rpm only asterisk install on CentOS 5.7. I'm using the asterisk repositories and I installed all the asterisk18 rpms, but find that chan_gtalk and res_jabber are missing. Is there a separate rpm that includes support for gtalk? Thanks in advance. -Gaurav -------------- next part -------------- An
2009 Sep 03
3
GTalk functionality Asterisk
Hello Previous context :- After Looking up sip and IAX2 that require configuration at router level which may cause some problems like connection break etc... so i left them ......... and start wondering if there is some thing that dont require configuration at router layer. The task to accomplish to make and recieve calls from outside local network using any protocol whose soft phones are
2009 Jan 29
2
GTalk Channel
Hello all, It used to work on calling my GTalk ID from another GTalk user. But now that I tried calling it again, the caller hears only a ringtone and disconnected after a few rings. The messages on my Asterisk-1.4.21.2 are the following: [Jan 29 10:37:51] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: Unexpected bind error: Cannot assign requested address [Jan 29 10:37:51] WARNING[1303]:
2007 Mar 03
1
gtalk2voip and Asterisk
hi, i was able to get this working with google talk. i entered myusername@gmail.com using the gtalk2voip.com website's "invite" box, and as a result, saw a request from service@gtalk2voip.com to be added as a buddy in my google talk contact list. i accepted the request. in my asterisk dialplan, i have this entry... exten => 3501, 1,
2009 Feb 04
1
Out of Office: Out of Office: Out of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Ou t of Office: Re : som e kind of t i meou t problem in pbx _sp ool.c
Me encuentro de vacaciones hasta el proximo 16/02. Por favor contactar a Pablo Minsteras (pablo.minsteras at utopixnetworks.com) o Gustavo Hernandez (gustavo.hernandez at utopixnetworks.com). Muchas Gracias, Gustavo Scheveloff
2008 Oct 25
1
gtalk dialstring?
Hi everyone! I couldn't find anything expressive about gtalk dialstrings. It doesn't seem to work. I'm not sure why, so I'll start at the easiest point. The syntax I found was: gtalk/my_account_name/buddys_account_name at gmail.com Is this correct? And does any of you googletalkers know, if a simple google-mail account is enough to use the talking bit, or do I have to
2011 May 13
1
asterisk 1.8 + google voice
somewhere along the way, i noticed incoming calls from google voice are no longer working on my asterisk 1.8.3.2 system. When the call comes in, asterisk immediately prints on the console: == Spawn extension (google-in, s, 2) exited non-zero on 'Gtalk/+12153930924-f947' [May 12 22:47:18] NOTICE[13835]: chan_gtalk.c:1977 gtalk_parser: Remote peer reported an error, trying to
2008 Feb 29
1
Gtalk with asterisk
Hi, I have been working with Asterisk for the ivr functionalities in the past. I am interested to implement the Jabber - Gtalk in asterisk. For which i installed the iksemel but this didnt help me out. I couldnt find the res_jabber.so file any where in the asterisk source directory. Still when i run the command "make menuselect" the channel driver "chan_gtalk" shows xxx
2010 Oct 25
4
google voice + asterisk: calls made to GV# processed but weird
Dear all, First off, I am very new to asterisk so forgive me if any of my comments or questions seem trivial. Thanks to [this post](http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/) and [this post](http://www.davidvossel.com/?p=28), I have GV set up on asterisk through jabber.conf and gtalk.conf. I can successfully dial out from asterisk. I'm trying to set up an
2007 Mar 07
0
gtalk2voip and Asteris
What kinds of problems were you having? I'm on 1.4.0 and chan_gtalk.so simply doesn't load. Of the 146 files in the /usr/lib/asterisk/modules/ directory, asterisk loads 144 of them, omitting only chan_gtalk.so and res_jabber.so. Connected to Asterisk 1.4.1 currently running on monkey (pid = 9371) Verbosity is at least 3 foo*CLI> module load chan_gtalk.so [Mar 7 10:23:07]
2007 Apr 01
1
No Audio with Gtalk
I configured my * with the instructions found here http://www.voip-info.org/wiki/view/Asterisk+Google+Talk to work with gtalk. The Phone rings and connects - but no audio! I am using a self-compiled asterisk 1.4.2 There is a lot of output on the CLI but I can't make sense of it. Perhaps somebody can help? Michael Output from the CLI: JABBER: gtalk_account OUTGOING: <iq
2012 Nov 02
3
Outgoing Google Motif Calls connect but continue ringing on outgoing side
I upgraded from Asterisk 10 to 11 and switched from gtalk.conf and jabber.conf to use motif.conf and xmpp.conf. I disabled gtalk and jabber from loading in modules.conf noload => res_jabber.so noload => chan_gtalk.so After copying my settings to the new conf files and restarting Asterisk I had no errors, but making outgoing calls from clients just kept ringing even though the other side
2012 Sep 25
2
undefined symbols
Hi, I compiled Asterisk 10.7.0 with gcc-4.5.3 and at runtime I'm getting these warnings: loader.c: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_smdi_interface_unref loader.c: Error loading module 'app_stack.so': /usr/lib/asterisk/modules/app_stack.so: undefined symbol: ast_agi_unregister loader.c: Error loading module
2012 Apr 01
0
10.3.0: gtalk_request: No XMPP client to talk to, us (partial JID)
Trying to use gtalk: -- Executing [andy at ipkall:2] Dial("SIP/ipkall-00000000", "gtalk/andy-gtalk/+1xxxyyyzzzz at voice.google.com") in new stack [Apr 1 10:41:53] ERROR[2416]: chan_gtalk.c:1934 gtalk_request: No XMPP client to talk to, us (partial JID) : andy-gtalk gtalk.conf [general] context=google-in ; Context to dump call into allowguest=yes stunaddr =
2011 Jun 25
3
How to export to pdf in landscape orientation?
Does anybody know how to get a pdf file with landscape orientation?. pdf(file= 'my_file.pdf' ,onefile=T,paper='A4') plot(sin, -pi, 2*pi) dev.off() Thank's in advance Juan A. Hernandez Spain [[alternative HTML version deleted]]
2011 Nov 18
2
Virtio-win RPM?
I should have this in my head by now... I tried migrating a physical Windows host last night using virt-p2v CD. It ran for 3 hours and then died. I'll get details later from the log but as I think about it, I'll bet it died because I never installed virtio-win in my Fedora migration server. The documentation says do "yum install virtio-win" - but this is RHEL
2010 Oct 07
0
Asterisk 1.8.0 Release Candidate 3 Now Available
The Asterisk Development Team has announced the third release candidate of Asterisk 1.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/. It is also very
2010 Oct 07
0
Asterisk 1.8.0 Release Candidate 3 Now Available
The Asterisk Development Team has announced the third release candidate of Asterisk 1.8.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/. It is also very