Displaying 20 results from an estimated 800 matches similar to: "10.3.0: gtalk_request: No XMPP client to talk to, us (partial JID)"
2011 Jan 06
0
using google for vm transcripts
I'm pretty impressed by how well (comparatively) google voice does in
doing voice mail transcripts. So I'd like to have google do my local
voice mail, and then email the transcript.
So I set up extensions.conf:
exten =>s,n,Dial(${House_Phones},36) ; this should be six rings
exten =>s,n,Dial(Gtalk/<my-user-name>/${<my-gv-number>{@voice.google.com)
but I get this
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
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]
2013 Apr 23
0
Asterisk 11.4.0-rc1 refuses to use the TURN server
After struggling with one way audio issues as a result of STUN binding
errors on both the Asterisk side and the Chrome side, we've decided to just
simply go with a TURN relay for RTP packets until the issues are resolved.
I configured rtp.conf so that all of the STUN related entries are commented
out, and I use the following TURN configuration instead:
turnaddr=numb.viagenie.ca:3478
;
;
2007 Aug 29
5
Ringing sound doesn't work
Hi,
I have these extensions:
exten => 101,1,Dial(SIP/101,15)
exten => 102,1,Dial(SIP/102,15)
exten => 0,1,Dial(SIP/101&SIP/102,15,r)
They work fine and I get the ringing sound if I dial them directly. However, I
also have this extension:
exten => s,1,Answer()
exten => s,2,Background(viagenie)
exten => s,3,WaitExten()
The ringing sound doesn't work for any extension
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
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,
2010 Oct 18
0
Asterisk 1.8.0 Release Candidate 4 Now Available
The Asterisk Development Team has announced the fourth release candidate of
Asterisk 1.8.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
With all currently 1.8.0 blocker issues closed, Asterisk 1.8.0-rc4 is currently
scheduled to become the full release of Asterisk 1.8.0.
All interested users of Asterisk are encouraged to
2010 Oct 18
0
Asterisk 1.8.0 Release Candidate 4 Now Available
The Asterisk Development Team has announced the fourth release candidate of
Asterisk 1.8.0. This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
With all currently 1.8.0 blocker issues closed, Asterisk 1.8.0-rc4 is currently
scheduled to become the full release of Asterisk 1.8.0.
All interested users of Asterisk are encouraged to
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]:
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
2006 Oct 17
1
One way audio on chan_gtalk
Hi!
I?m trying with 1.4b2, chan_jabber and
chan_gtalk. Jabber client register fine on
talk.google.com, and when I start a call from
gtalk to asterisk, I can see the incoming call
and I see that asterisk play prompts (ie: demo
and thank-you), but i can?t hear audio. If I
redirect incoming call to a sip client, at sip I
can hear but I can't in google talk.
Asterisk is at public no
2008 Sep 15
0
rc6: Dunno what to do with STUN message 0101 ??
Having some trouble with sip behind a nat. So tried:
stunaddr = numb.viagenie.ca
in sip.conf. Didn't help so tried stun debug:
asterisk*CLI> stun set debug on
STUN Debugging Enabled
STUN Packet, msg Binding Response (0101), length: 36
Found STUN Attribute Mapped Address (0001), length 8
Ignoring STUN attribute Mapped Address (0001), length 8
Found STUN Attribute Changed Address (0005),
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
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
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
2009 Nov 30
0
Gtalk Asterisk integration
Hello users,
I am trying to integrate asterisk and gtalk.
my configuration is as follows
OS:centos
asterisk-1.6.0
asterisk-addons-1.6.0
dahdi-linux-2.2
dahdi-tools-2.2
libpri-1.4 share
iksemel-1.2
#/etc/asterisk/jabber.conf
[general]
debug=yes
autoprune=no
autoregister=no
[google]
type=client
serverhost=talk.google.com
username=XXXX at gmail.com
secret=xxxxx
port=5222
usetls=yes
usesasl=yes
2007 Jun 21
3
gtalk - no audio
Hi list,
I'm trying to get channel gtalk working in asterisk 1.4.5
I have it built and configured as follows:
*jabber.conf:*
[general]
debug=yes
autoprune=no
autoregister=no
[myaccount]
type=client
serverhost=talk.google.com
username=myaccount at gmail.com/Talk
secret=mypassword
port=5222
usetls=yes
usesasl=yes
statusmessage="Talk to me"
timeout=100
*gtalk.conf:*
[general]