similar to: GTalk/Jabber passing audio in 1.4.1!

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

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

I just upgraded a box to 1.2.3 this morning after encountering the issues noted earlier on the list. Everything is great. In fact, a LOT better. In the past few weeks, I've been battling with BV to address dropped outgoing voice packets (the flipside is that I haven't experienced this with other providers during tests), and an annoying mechnical 'chirp' at the start of a call.

I've just added a PREVIEW release of my upcoming how-to guide for Asterisk PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2. It addresses all kinks and showstoppers that many people have experienced over the past year or so. Because this is a preview, it is not the final version of this guide. It is subject to change (format, copy, layout, etc.) To view and download

-----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 :

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

After months of BroadVoice ignoring my trouble tickets for dropped calls, delayed termination, etc., I'm throwing in the towel. While they have credited $19.95 to my account, they refuse to credit anything more, despite ALL of the problems I've had. I feel the least they could do is credit the remaining $8.61 to my account, yet they won't. I haven't really been following up on

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 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

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,

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

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]:

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="

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

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

Hi! I just tried to call a friend using jingle, but I got refused. Errorcode was 502, he tried to call me, heard it ringing once and then it stopped. I used: originate jingle/gtalk_account/friend at jabber.linuxlovers.at [application] I'm registered to googletalk, but this should mean no harm, or should it. Once I was able to receive a text-message from him, but couldn't

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

JABBER: gtalk_account OUTGOING: <?xml version='1.0'?><stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to=' gmail.com' version='1.0'> localhost*CLI> jabber show tes JABBER: gtalk_account INCOMING: <?xml version="1.0" encoding="UTF-8"?><stream:stream from="gmail.com"

Hello everyone! Philippe, you told me to make a bugreport. Well, here it comes, I'm still not sure, if tis is a bug or a miss-configuration. So I've put up a collection of configurations/output/debug files from a simple asterisk session testing the gtalk call. You can download it here: http://juliencoder.de/ap.txt Or I can mail it, just tell me where and I'll attach it to

I was optimizing a 650MB using ferret (0.11.3) and I received the following error. I''ve seen some people have similar issues but I haven''t seen any resolutions. The contents of the index directory follow the error. Has anyone seen anything like this and found a resolution? Many thanks. /mnt/apps/search/releases/20070622175637/script/../config/../vendor/