[asterisk-users] Gtalk/Jabber Issue
Vladimir Mikhelson
vlad at mikhelson.com
Fri Feb 11 01:41:23 CST 2011
William,
Another thing to exclude is networking. Can you verify that nothing
blocks the specific traffic on your network? Any chance of taking the
packet trace on your gateway?
-Vladimir
On 2/11/2011 1:18 AM, William Stillwell wrote:
>
> I don’t’ appear to have an jabber [] OUTGOING packets?
>
>
>
> I get just 1 incoming packet, and it just sits there, until it rings
> to voicemail.
>
>
>
>
>
>
>
> *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 1:47 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Gtalk/Jabber Issue
>
>
>
> 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>
> [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.
>
>
>
>
>
>
>
>
>
> --
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>
> --
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>
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> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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