similar to: Gtalk Asterisk integration

Hi all, Do some one experiencing running jabber applications (jabberstatus...) in asterisk? I do experinced Asterisk 1.4.18 and wish to start it, however I got such result. IBM*CLI> help jabber No such command 'jabber'. IBM*CLI> help jabberstatus No such command 'jabberstatus'. Any one can help me on this, or may be I miss out somethings that cause jabber applications

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 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 all, Do some one experiencing running jabber applications (jabberstatus...) in asterisk? I do experinced Asterisk 1.4.18 and wish to start it, however I got such result. IBM*CLI> help jabber No such command 'jabber'. IBM*CLI> help jabberstatus No such command 'jabberstatus'. Any one can help me on this, or may be I miss out somethings that cause jabber applications

Hi Lee, You need to specify the VM context that you are using.. so using your examples.. extensions.conf entry.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000) exten => 1000,102,Voicemail2(b1000) exten => 1000,103,Hangup should be.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000@sip) exten => 1000,102,Voicemail2(b1000@sip) exten

Hi, I'm running dsync migrations using imapc and the source IMAP server is just too slow. It has taken 8 hours to migrate a mailbox with 47,000 messages. It seems most of the mailboxes are never accessed with IMAP, so it would be just as good to migrate them via POP3 which should work faster in my case. Is this actually meant to work? doveadm -D -o pop3c_user=xxxxxx at example.com -o

Hi, I've got a relatively simple tinc setup. I've got two "servers" that are on the public internet that act as routers for three "clients" that are behind NATs. Those servers are called aaaaa and bbbbb the clients are xxxxx, yyyyy and zzzzz Unfortunatly the servers have problems accepting a connection from the clients syslog on aaaaa: Sep 29 18:28:58 schuerrer

I have been doing some reading about gtalk and asterisk. Most of it is pointed to enable using gtalk for making phonecalls. Would it be possible to use gtalk instant messaging input (just some text send to the gtalk account configured on an asterisk box) into the dialplan. This way you could use gtalk im to trigger all kind of events like sending an sms, adding sip entries to the system,

Hi, i've just finished setting up gtalk connection with asterisk. it works nice, audio is full duplex. i just have one question which i could not find an exact answer to. Is gtalk able to send dtmf codes? Because i'd like to listen to my voicemails while away from home. I've been googling for half an hour, i found some sort of jingle protocol which i'm not sure what to

Hello everyone! I just installed the latest asterisk from svn. Now I'm retrying my luck with gtalk and jingle. I have moved so my basic setup has changed a bit... I'm not sure if it helps or hurts. I tried this: call myself: channel originate gtalk/gtalk_account/juliencoder at googlemail.com application \ Jack i(system:playback_1)o(system:capture_1) I got some notes about a lot

Hello everyone! I'm just trying to set up my new asterisk (version 1.6.2.8-rc1). I'd be very grateful, if someone could help me here. I'd be very glad, if one of you could test googletalk with me. Last time I tried (in 1.6.0.x times) it wouldn't work in the end. But here are my gtalk and jabber.conf files. Could you please take a look and tell me, if the settings same sane?

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]

I'm working with a SIP provider to try and transition our sip connection with them to PJSIP. I thought I had transitioned the settings correctly, but whenever I attempt an Originate it never even tries to send any PJSIP messages. I'm currently running Asterisk 13.0.0. Anyone have any suggestions as to what I am doing wrong? The SIP provider says the latest version of Asterisk they have

hi all, i have a centos 5.4 64 bit system running on the base hardware of my system and have windows machines running in virtual in vmware workstation. i need something to *voice chat* with *GTALK* users. In the windows VMs i can install gtalk for windows which does support voice chat, but the problem there is the microphone doesnt get detected inside the VMs. incase of centos installed on the

Hello everyone! So I hacked app_jack.c today, as best I could. Whic came mostly down to inserting ast_log() messages. I discovered the following with JACK: When it starts, it tries to read 512 bytes and only gets 0. That clears up after a while. Sometimes a good time later than the reading comes the writing. And there the real strangeness might begin. Because usually the framebuffer

Hi all, I'm trying to establish jingle call in this network scenario: Asterisk -> NAT -> Internet -> HTTP_PROXY -> GTalk client The call is received and answered in gtalk but there is no audio in the call. I suppose it could be related to the support for relay candidates in asterisk jingle implementation. Anybody else has faced this problem? Notes: Asterisk 1.6.2. It

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 run a sshd host solely to allow employees to tunnel secure connections to our internal hosts. Some of which do not support encrypted protocols. These connections are chroot'ed via the following in /etc/ssh/sshd_config Match Group !wheel,!xxxxxx,yyyyy AllowTcpForwarding yes ChrootDirectory /home/yyyyy X11Forwarding yes Where external users belong to group yyyyy (primary). We

Hello. I would like to configure Asterisk to accept incoming calls from two different GoogleVoice numbers via gtalk and jabber. I'm running Asterisk 1.8.3.2 and I can get one number working just fine. However, I can't figure out how to modify the gtalk.conf file shown on the Asterisk wiki site to work with two different jabber profiles. Do all incoming GoogleVoice calls have to go

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! Are hints not yet implemented in res_jabber? I have this here: exten => 3000,hint,gtalk/gtalk_account/mari.xxxxxxx at gmail.com But the hint doesn't show any difference. It always shows online on the phone and core show hints always shows that: 6003 at internal : SCCP/6003 State:Unavailable Watchers 0 6002 at internal :