[asterisk-users] Gtalk/Jabber Issue

A E [Gmail] all.eforums at gmail.com
Mon Jul 18 23:02:11 CDT 2011


On Mon, Feb 21, 2011 at 1:21 AM, Vladimir Mikhelson <vlad at mikhelson.com>wrote:

> William,
>
> It still looks like something is not properly set with your account on
> Google Voice.  Have you had a chance to follow the recommendations I
> gave you earlier in the thread?
>
> If the account is properly set the dial string will need to look like
> this,  "gtalk/<jabber-conf-section-name>/+$OUTNUM$@voice.google.com"
> where $OUTNUM$ is a called number in the international format.
>
> On the receiving end the call will come with an empty CID Number, but
> with the CID Name which looks like this:
> +15555551212 at voice.google.com/srvres-MTAuMjE4LjIuMTk3Ojk4MzM=
>
> Just cut all prior to "@" as a CID Number. See
> https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google
>
> Also you do not need to wait 5 seconds. 1 or 2 is sufficient.
>
> -Vladimir
>
>
> This is a really old thread but I am having the same issues as William was
having. The incoming call just doesn't hit the context in extensions.conf. I
see the call come in on jabber...but I've tried almost 4-5 different
variations of handling the call in extensions.conf from examples on the web,
but nothing happens. I'm on 1.8.5.0.

BTW, there is no Google Voice involved. and I'm calling from from a gmail
based gtalk client. Also, I can successfully make an outbound call. Just the
inbound isn't working :( Any help please?

Currently my incoming dial-plan is:
[gtalk-in]
exten => s,1,Answer()
        same => n,Wait(2)
        same => n,SendDTMF(1)
        same => n,Dial(SIP/2000,20)

and I have tried a whole bunch of stuff in jabber.conf and gtalk.conf but
nothing seems to cut it. I have also tried using matching my email address
(called gtalk a/c) to match in the exten as opposed to 's' extension and
that doesn't work either.

gtalk.conf
--------------
[general]
context=gtalk-in
bindaddr=0.0.0.0
externip=<my external address>
allowguest=yes

[guest]
disallow=all
allow=ulaw
context=gtalk-in
connection=asterisk

[aeg74]
username=aeg74 at gmail.com
disallow=all
allow=ulaw
context=gtalk-in
connection=asterisk

jabber.conf
============
[general]
debug=yes
autoprune=yes
autoregister=yes

[asterisk]
type=client
serverhost=talk.google.com
username=all.eforums at gmail.com/Talk
secret=<my secret>
port=5222                               ; Port to use defaults to 5222
usetls=yes                              ; Use tls or not
usesasl=yes                             ; Use sasl or not
buddy=aeg74 at gmail.com
status=available
statusmessage="On Asterisk"
timeout=100

*This is the debug on jabber*
JABBER: asterisk INCOMING: <iq type="set" to="
all.eforums at gmail.com/Talk17BFE21F" id="CA051C15DD949454" from="
aeg74 at gmail.com/gmail.320B5151"><jin:jingle action="session-initiate"
sid="c1901211999" initiator="aeg74 at gmail.com/gmail.320B5151"
xmlns:jin="urn:xmpp:jingle:1"><jin:content name="audio"
creator="initiator"><rtp:description media="audio"
xmlns:rtp="urn:xmpp:jingle:apps:rtp:1"><rtp:payload-type id="103"
name="ISAC" clockrate="16000"><rtp:parameter name="bitrate"
value="32000"/></rtp:payload-type><rtp:payload-type id="104" name="ISAC"
clockrate="32000"><rtp:parameter name="bitrate"
value="56000"/></rtp:payload-type><rtp:payload-type id="119" name="ISACLC"
clockrate="16000"><rtp:parameter name="bitrate"
value="40000"/></rtp:payload-type><rtp:payload-type id="99" name="speex"
clockrate="16000"><rtp:parameter name="bitrate"
value="22000"/></rtp:payload-type><rtp:payload-type id="97" name="IPCMWB"
clockrate="16000"><rtp:parameter name="bitrate"
value="80000"/></rtp:payload-type><rtp:payload-type id="9" name="G722"
[Jul 18 23:36:15]
JABBER: asterisk INCOMING: clockrate="16000"><rtp:parameter name="bitrate"
value="64000"/></rtp:payload-type><rtp:payload-type id="102" name="iLBC"
clockrate="8000"><rtp:parameter name="bitrate"
value="13300"/></rtp:payload-type><rtp:payload-type id="98" name="speex"
clockrate="8000"><rtp:parameter name="bitrate"
value="11000"/></rtp:payload-type><rtp:payload-type id="3" name="GSM"
clockrate="8000"><rtp:parameter name="bitrate"
value="13200"/></rtp:payload-type><rtp:payload-type id="100" name="EG711U"
clockrate="8000"><rtp:parameter name="bitrate"
value="64000"/></rtp:payload-type><rtp:payload-type id="101" name="EG711A"
clockrate="8000"><rtp:parameter name="bitrate"
value="64000"/></rtp:payload-type><rtp:payload-type id="0" name="PCMU"
clockrate="8000"><rtp:parameter name="bitrate"
value="64000"/></rtp:payload-type><rtp:payload-type id="8" name="PCMA"
clockrate="8000"><rtp:parameter name="bitrate"
value="64000"/></rtp:payload-type><rtp:payload-type id="117" name="red"
clockrate="8000"/><rtp:payload-type id="106" name="
[Jul 18 23:36:15]
JABBER: asterisk INCOMING: telephone-event"
clockrate="8000"/></rtp:description><p:transport xmlns:p="
http://www.google.com/transport/p2p"/></jin:content></jin:jingle><ses:session
type="initiate" id="c1901211999" initiator="aeg74 at gmail.com/gmail.320B5151"
xmlns:ses="http://www.google.com/session"><pho:description xmlns:pho="
http://www.google.com/session/phone"><pho:payload-type id="103" name="ISAC"
bitrate="32000" clockrate="16000"/><pho:payload-type id="104" name="ISAC"
bitrate="56000" clockrate="32000"/><pho:payload-type id="119" name="ISACLC"
bitrate="40000" clockrate="16000"/><pho:payload-type id="99" name="speex"
bitrate="22000" clockrate="16000"/><pho:payload-type id="97" name="IPCMWB"
bitrate="80000" clockrate="16000"/><pho:payload-type id="9" name="G722"
bitrate="64000" clockrate="16000"/><pho:payload-type id="102" name="iLBC"
bitrate="13300" clockrate="8000"/><pho:payload-type id="98" name="speex"
bitrate="11000" clockrate="8000"/><pho:payload-type id="3" name="GSM"
bitrate="13200" clockrate="8000"/><pho:
[Jul 18 23:36:15]
JABBER: asterisk INCOMING: payload-type id="100" name="EG711U"
bitrate="64000" clockrate="8000"/><pho:payload-type id="101" name="EG711A"
bitrate="64000" clockrate="8000"/><pho:payload-type id="0" name="PCMU"
bitrate="64000" clockrate="8000"/><pho:payload-type id="8" name="PCMA"
bitrate="64000" clockrate="8000"/><pho:payload-type id="117" name="red"
clockrate="8000"/><pho:payload-type id="106" name="telephone-event"
clockrate="8000"/></pho:description></ses:session></iq>
[Jul 18 23:36:37]
JABBER: asterisk INCOMING: <iq type="set" to="
all.eforums at gmail.com/Talk17BFE21F" id="415FBBCB2085F931" from="
aeg74 at gmail.com/gmail.320B5151"><jin:jingle action="session-terminate"
sid="c1901211999" xmlns:jin="urn:xmpp:jingle:1"><ses:reason xmlns:ses="
http://www.google.com/session"><ses:connectivity-error/></ses:reason><pho:call-ended
xmlns:pho="http://www.google.com/session/phone"/></jin:jingle><ses:session
type="terminate" id="c1901211999" initiator="aeg74 at gmail.com/gmail.320B5151"
xmlns:ses="http://www.google.com/session"><ses:reason><ses:connectivity-error/></ses:reason><pho:call-ended
xmlns:pho="http://www.google.com/session/phone"/></ses:session></iq>

Thanks in advance
aeg
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110719/2c4082d3/attachment.htm>


More information about the asterisk-users mailing list