similar to: chan_gtalk load error

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

hi, it turns out that the iksemel library (which i installed using an rpm) was returning 0 when the function iks_has_tls() was called. it should return 1 otherwise res_jabber.o thinks gnuTLS is not installed. i confirmed this by running a test program i wrote, that calls iks_has_tls . it returned 0. i downloaded iksemel source, compiled it and now the test program returned 1. now,

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]

I have installed gnutls and gnutls-devel from RedHat repositories [root at asterisk asterisk]# yum install gnutls gnutls-devel I have installed iksemel with gnutls support : [root at asterisk asterisk]# cd /usr/src/iksemel-1.3/ [root at asterisk asterisk]# ./configure --with-gnutls --prefix=/usr [root at asterisk asterisk]# make [root at asterisk asterisk]# make check [root at asterisk

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

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

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

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]:

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,

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

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

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

I rebooted our server Fedora 17 today, and now asterisk won't start; asterisk[1063]: segfault at 0 ip 00007f117aee122d sp 00007fffbc398990 error 4 in libiksemel.so.3.1.1[7f117aed8000+d000] iksemel is required for motif and xmpp. I downloaded the iksemel source and rebuilt. No luck. Any help really appreciated. sean

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

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

Hi All I am developing a client that uses libjingle to do xmpp stuff with ejabberd. I can also make audio calls between those clients. What I am trying to archive now is to send calls to pstn using jingle. I was told in the jingle-dev community that asterisk can do that. Is there any way to send jingle audio calls to asterisk and will it understand them ? If yes..can I forward those calls to PSTN

Hello I'm curious about what can be done when using Jabber with Asterisk. What are good examples of this combination? Thanks.

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

I've just compiled Asterisk 1.4.1 and I'm happy to report that I've got two-way audio between Google Talk and Asterisk! This IS an exciting moment today in VoIP! This is just GREAT! - Ronald Lewis http://ronaldlewis.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am trying to drop rows of a dataframe based on values of the column PID, but my strategy is not working. I hope someoen can tell me what I am doing incorrectly. # Values of PID column > jdata[,"PID"] [1] 16608 16613 16355 16378 16371 16280 16211 16169 16025 11595 15883 15682 15617 15615 15212 14862 16539 [18] 12063 16755 16720 16400 16257 16209 16200 16144 11598 13594 15419 15589