similar to: Gtalk with asterisk

Hi, I use asterisk 1.4.11 version for making outbound calls. Running it on linux(fedora core 7) machine. Recently purchased the g729 codec, got it registered with my asterisk box. I have two queries for you to help me. 1. How do i know when an outbound call is placed that it makes use of the g729 codec. when i use the command "show g729" i get the following: 0/0 encoders/decoders of

Hi, I have been working with asterisk to make ivr calls (outbound and inbound). I have the functionality - Read(variable|file_name) used in my dialplan. Now i need to pass the variable to my ruby file to compare the data entered with the database (mysql). How can i pass the arguments from my dialplan to the ruby file. Is there a way i can do it with the agi script? Any one has any clues on

Hi, Was just wondering about the bug fix in asterisk-1.6 Beta Release.. Its been said that "Added an 'n' option to SpeechBackground to request that the channel not get answered". Can anyone brief me about what exactly this bug are they talking about. We had issues with Background syntax working in macro. Is this the bug fix they are talking about. Regards, Naveen.Palani

Hi, I have the asterisk-1.4.11 set up installation on my Ubuntu machine. When i try making a simple incoming call using xlite softphone. I get the following message when i try calling to the number. *CLI> [Feb 19 13:35:40] NOTICE[4137]: chan_sip.c:5331 process_sdp: No compatible codecs, not accepting this offer! Which codec is it talking abt here. How can i resolve this. My dialplan is as

Hi, I am using Asterisk-1.4.11 version to make outbound calls and deliver the swift text to audio. My functionality is as for example i make this text to audio deliver the person called. Eg. swift -o /tmp/test.wav -p audio/channels=1,audio/sampling-rate=8000 "Press 1 to confirm. Press 3 to cancel." extension.conf dialplan: [dialout] exten =>

Hi, I have been using asterisk-1.4.17 version. Have a SIP registration from bandtel sip providers. Use DID numbers for the incoming calls which works fine when i dont use any peer setting in my sip.conf file. But when i use a peer and make calls thru the DID number it doesn't reach asterisk at all. Doesnt give me any errors as well. peer in my sip conf is as given below: [proxy2_bandtel]

Hello, I am wanting to close a specific channel for example; SofthangUp(SIP/EXTEN-UNIQUEID) but the problem is the channel is assigned a unique id as well. The need fits into the idea of receiving a call from a higher status user and thus closing a specific channel to allow the higher priority call to route through the dial plan to the freed extension. Any ideas welcome. Many thanks

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, 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,

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

I'm working from the docs here: http://voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk and getting an error doing the ./configure on the iksemel module: checking for getaddrinfo... yes ./configure: line 20399: syntax error near unexpected token `,' ./configure: line 20399: `AM_PATH_LIBGNUTLS(,' It seems to want the "libgnutls-dev" package as per the

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

I want to use JabberSend in my dialplan, but I saw that my Asterisk does not support Jabber. Also I have nowhere a module res_jabber.so... So I thought I'd rebuild my Asterisk. In menuselect I saw that res_jabber was dependent of 'iksemel' and 'gnutls'. In my yum repositories I can find a gnutls.i386, but what is this iksemel-beast ??? There is info to find via google on

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

OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can't seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes

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

I've included my jabber.conf below. I'm betting the following errors: [Jul 18 21:05:22] ERROR[28166]: res_jabber.c:609 aji_act_hook: JABBER: Node Error [Jul 18 21:05:22] WARNING[28166]: res_jabber.c:1537 aji_recv_loop: JABBER: Got hook event. jabber test [Jul 18 21:04:16] WARNING[32691]: res_jabber.c:1421 ast_aji_send: JABBER: Not connected can't send User: bferrell at gtalk.com

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

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]: