similar to: asterisk a "less secure app" on google ??

Hi all, Been reading about chan_motif / chan_xmpp in the wiki's for 1.8, 10 and 11 version of asterisk. In each example i got the impression that the asterisk server is registering on a XMPP server as a single user with the credentials as specified in jabber.conf. Instead of a single xmpp-user, could that also be multiple users? For instance, for each sip-user an xmpp-user? When i skim

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 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]

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="

When a caller calls my google voice phone number, I must answer, wait and press one to accept. Sometimes even that does not work. I have tried a few different things to get asterisk to place the call in an answered state and send the DTMF 1 with the Dial macro. I found Malcom Davenports wiki page regarding Google calling which has been very helpful in troubleshooting the issue.

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:

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, has anyone managed to establish an XMPP connection to the facebook Jabber servers? I'd like to send messages on missed calls vie FB. - -S - -- (o_ Stefan Gofferje | SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler & Koch - the original point and click interface -----BEGIN PGP

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

Trying to use gtalk: -- Executing [andy at ipkall:2] Dial("SIP/ipkall-00000000", "gtalk/andy-gtalk/+1xxxyyyzzzz at voice.google.com") in new stack [Apr 1 10:41:53] ERROR[2416]: chan_gtalk.c:1934 gtalk_request: No XMPP client to talk to, us (partial JID) : andy-gtalk gtalk.conf [general] context=google-in ; Context to dump call into allowguest=yes stunaddr =

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?

Hello, I am attempting to setup Asterisk to work with Gtalk. I am using the following versions: Slackware Linux 12.0 Asterisk 1.6.2.9 GNU TLS 2.8.6 Iksemel (svn v25) OpenSSL 0.9.8o It all compiles however about 10 seconds after starting Asterisk it crashes. To mitigate this issue I have moved jabber.conf to another directory and then Asterisk starts up. So I assume the issue is with this

I've set up google voice to chat with me: Forwards calls to: <me>@gmail.com and xmpp: [general] debug=no ; Enable debugging (disabled by default). autoprune=yes ; Auto remove users from buddy list. Depending on your ; setup (ie, using your personal Gtalk account for a test)

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

I'm trying to get my Asterisk 11 test box set up with XMPP, having troubles with JabberSend(). My jabber.conf file is as follows: [general] debug=no autoprune=no [testaccount] type=client serverhost=my.jabber.server username=myaccount at my.jabber.server secret=mypassword port=jabberport usetls=yes usesasl=yes xmpp show connections gives the following output from the console:

Hi all, Iam using asterik 1.4.8 and connected to google talk. When iam calling from my google talk account to sip phone i can hear the voice (2 way). (this happens only within the LAN). when my friend tries to call my asterisk server (connects to the public ip) using his googletalk client it comes to my sip phone but either party cant hear a voice. I have fully allowd both tcp,udp on my

On 11.9, trying to set up a callcentric peer: sip debug: > <--- SIP read from UDP:204.11.192.161:5060 ---> > INVITE sip:1777<myccid>@10.10.11.180:5060 SIP/2.0 > v: SIP/2.0/UDP 204.11.192.161:5060;branch=z9hG4bK-6104e46aaaaef4249814d16a2ffb990d > f: <sip:<calling number>@66.193.176.35>;tag=3606475083-968127 > t:

I've been getting a lot of timeouts on non-critical invite transactions. I turned on sip debug. They were the result of SIP invites like this: Retransmitting #10 (NAT) to 185.107.94.10:13057: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 215.45.145.211:5060;branch=z9hG4bK-524287-1---zg4cfkl50hpwpv4p;received=185.107.94.10;rport=13057 From:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom