[asterisk-users] Gtalk/Jabber Issue
Vladimir Mikhelson
vlad at mikhelson.com
Fri Feb 11 00:47:13 CST 2011
William,
I have gone through the similar frustration recently. Everything works
as of early morning yesterday. The big difference, I am on 1.8.2.3.
Have you seen this ticket on the tracker
https://issues.asterisk.org/view.php?id=10512 ? Anything applicable to
your case? The messages are identical to yours on the outgoing call.
-Vladimir
On 2/11/2011 12:32 AM, William Stillwell wrote:
>
> Still no dice..
>
>
>
> This make no since.. ive gone over the config a million times now..
>
>
>
> The windows gtalk /voice client works just fine. (incoming and
> outgoing calls)
>
>
>
>
>
>
>
> *From:*asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of
> *Vladimir Mikhelson
> *Sent:* Friday, February 11, 2011 12:51 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Gtalk/Jabber Issue
>
>
>
> William,
>
> I have just noticed that you have several configuration statements
> commented out.
>
> I would suggest to un-comment the "status=" in jabber.conf. I would
> also suggest to un-comment the "timeout=", I am not that concerned of
> the "keepalive=".
>
> You can reload jabber, no need to restart the Asterisk.
>
> -Vladimir
>
>
>
> On 2/10/2011 11:40 PM, Vladimir Mikhelson wrote:
>
> William,
>
> Have you tried outgoing calls? What happens there?
>
> Have you restarted the Asterisk after you fixed the typo?
>
> -Vladimir
>
>
>
> On 2/10/2011 10:44 PM, William Stillwell wrote:
>
> Yeah, that was a typo, but I fixed, still no dice.
>
>
>
> The incoming jabber call doesn’t fire the gtalk connection.
>
>
>
>
>
> *From:*asterisk-users-bounces at lists.digium.com
> <mailto:asterisk-users-bounces at lists.digium.com>
> [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of *Warren
> Selby
> *Sent:* Thursday, February 10, 2011 10:16 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Gtalk/Jabber Issue
>
>
>
> You've got connection=jp_jabber defined in one file, and [jb_jabber]
> defined in the other.
>
> Thanks,
>
> --Warren Selby, dCAP
>
>
> On Feb 10, 2011, at 5:55 PM, "William Stillwell"
> <william at stillwellsoft.com <mailto:william at stillwellsoft.com>> wrote:
>
> Sorry, Asterisk Build 1.6.2.7
>
>
>
> *From:*asterisk-users-bounces at lists.digium.com
> <mailto:asterisk-users-bounces at lists.digium.com>
> [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of
> *William Stillwell
> *Sent:* Thursday, February 10, 2011 6:50 PM
> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
> *Subject:* [asterisk-users] Gtalk/Jabber Issue
>
>
>
> 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
>
> ;autoprune=no
>
> autoregister=yes
>
>
>
>
>
> [jb_jabber]
>
> type=client
>
> serverhost=talk.google.com
>
> username=XXXXXXXXX at gmail.com
> <mailto:username=XXXXXXXXX at gmail.com>/Talk
>
> secret=XXXXXXX
>
> port=5222
>
> usetls=yes
>
> usesasl=yes
>
> ;status=Available
>
> statusmessage="Connected via Asterisk"
>
> ;timeout=100
>
> ;keepalive=yes
>
>
>
>
>
> Extensions.conf
>
>
>
> [google-in]
>
> exten => s,1,NoOp(Call from GTalk)
>
> exten => s,n,Set(CallerID(Name)="From GoogleTalk")
>
> exten => s,n,Dial(SIP/1000)
>
>
>
> jabber show connected
>
>
>
> Jabber Users and their status:
>
> User: xxxxxx at gmail.com <mailto:xxxxxx at gmail.com>/Talk -
> Connected
>
> ----
>
> Number of users: 1
>
>
>
>
>
> ---- CLI on incoming Call ----
>
>
>
> bannana*CLI>
>
> JABBER: jb_jabber INCOMING: <iq
> from="+1*********@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2
> <mailto:+1*********@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2>"
> to="******@gmail.com/TalkD876FAA0
> <mailto:******@gmail.com/TalkD876FAA0>"
> id="jingle:10.218.14.137-17447266:1:03800E94"
> type="set"><ses:session type="initiate"
> id="SIP1007753261 at 10.218.122.83
> <mailto:SIP1007753261 at 10.218.122.83>"
> initiator="+1*******@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2
> <mailto:+1*******@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2>"
> xmlns:ses="http://www.google.com/session"><pho:description
> xmlns:pho="http://www.google.com/session/phone"><pho:payload-type
> id="0" name="PCMU" clockrate="8000"/><pho:payload-type id="101"
> name="telephone-event"/></pho:description><transport
> behind-symmetric-nat="false"
> can-receive-from-symmetric-nat="false"
> xmlns="http://www.google.com/transport/raw-udp"/><transport
> xmlns="http://www.google.com/transport/p2p"/></ses:session></iq>
>
> bannana*CLI>
>
> JABBER: jb_jabber INCOMING: <iq
> from="+1********@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2
> <mailto:+1********@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2>"
> to="******@gmail.com/TalkD876FAA0
> <mailto:******@gmail.com/TalkD876FAA0>"
> id="jingle:10.218.14.137-17447266:1:03800EB9"
> type="set"><ses:session type="terminate"
> id="SIP1007753261 at 10.218.122.83
> <mailto:SIP1007753261 at 10.218.122.83>"
> initiator="+1*******@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2
> <mailto:+1*******@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2>"
> xmlns:ses="http://www.google.com/session"><pho:call-ended
> xmlns:pho="http://www.google.com/session/phone">Call
> cancelled</pho:call-ended></ses:session></iq>
>
> bannana*CLI>
>
>
>
>
>
> it doesn’t even try to fire the google-in context ?
>
>
>
> Lastest Version of iksemel Installed, asterisk was rebuild after
> installed, asterisk sees both jabber/gtalk commands.
>
>
>
> It just will NOT ring my dialplan.
>
>
>
>
>
>
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110211/9bcd7f3e/attachment.htm>
More information about the asterisk-users
mailing list