similar to: GTalk Channel

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,

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 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

Hello folks. I've got a strange issue. When I modprobe TE220 I do not see mesages like Launching card: 0 <..> Setting up global serial parameters. You can see how I loaded and unloaded the card for several times - http://asteriskpbx.ru/pastebin/11 lspci can detect the card: 03:08.0 Communication controller: Digium, Inc. Device 0220 (rev 02) dahdi_hardware also: astpbx ~ # dahdi_hardware

Hi I have a table read from a mysql database which is of the kind clusterid clockrate I obtained this table in R as clockrates_table <-sqlQuery(channel,"select...."); I have a function within which I wish to extract the clusterid for a given cluster. Although I know that there is just one row per clusterid in the data frame, I am using subset to extract the clockrate. clockrate =

Hi I have a table that goes data cluster_ac clockrate age class 7337 0.9 0.001 alpha_proteins 7888 0.1 0.78 beta proteins etc The class column can have 7-8 different unique values While the clockrate and age columns are floats varying from 0 to 1. I wish to get the average clockrate across each of the classes for this data. I would appreciate your help

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

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

Hello all, I am running Asterisk-1.4.5 on my Debian GNU/Linux Etch here and I want to send the voicemails as attachment to e-mails and delete the voicemails from my PBX once it has been sent. But, I don't have a running MTA here even on the PBX itself. I just want to send the e-mails to my GMail account from my PBX. Can I just use the mail or mailx command to send the e-mail and attach the

Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. This one way audio

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

I'm failing to understand how getrusage() works, which is a bit perplexing, because it doesn't seem like it would be terribly complicated. I've attached the code. My aim is to verify that I can use getrusage() to do (admittedly crude) instrumentation of which functions in my program are allocating lots of memory[1]. So I figure I can call getrusage() at various points and look at

I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ... I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly

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

Hello all, Where can I find a list of description for each sound files provided by the asterisk-sounds-main Debian package? You can find the contents of my /usr/share/asterisk/sounds/ directory at http://paste.debian.net/33679. Thank you in advance. GNUbie -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I'm planning to create a simple Auto-Attendant (IVR Menu) for my home PBX yet all callers from incoming (trunk) calls must only press the extension numbers from the [analog-ext] else will play the "pbx-invalid". How do you do that using the GotoIf() (or probably using the other applications) but will check if the numbers entered belongs to a specific context? Also, how

I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from

Hello all, Is it possible to store, read and write configuration files in an SQLite3 database instead of using the configuration files inside the /etc/asterisk/ directory? If it is then can you point me to the right documentation on how to do this or probably hints on how to do this? Thank you in advance. GNUbie -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hello, I have a Google Voice phone number and want to connect it to my asterisk box to have calls handled to my SIP account. When I call the number I receive the correct INCOMING request on Jabber portion of asterisk, but the call is not connected to the gtalk part. JABBER: asterisk INCOMING: <iq from="+ 17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" to="