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[asterisk-users] chan_motif / res_xmpp problems


 
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daniel at pocock.pro
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PostPosted: Mon Jul 21, 2014 4:41 am    Post subject: [asterisk-users] chan_motif / res_xmpp problems Reply with quote

I've now replicated my setup on a host with a single IPv4 address and I
am still having trouble with the ICE negotiation.

I am trying to call from Jitsi to Asterisk through a Prosody XMPP
server. Asterisk successfully registers with the XMPP server and
appears to be available in the buddy list in Jitsi. Jitsi is being run
with the "-4" command line option to use IPv4 only just in case Asterisk
doesn't like to see IPv6 ICE candidates.

I try clicking to make an audio-only call from Jitsi. In the Asterisk
logging (xmpp set debug on) I see the incoming "session-initiate" XML
stanza but Asterisk does not send any XML back.

I definitely have "icesupport=yes" in rtp.conf and I've tried it with
and without specifying a TURN server from each end.

Is this working for anybody?


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jcolp at digium.com
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PostPosted: Mon Jul 21, 2014 7:34 am    Post subject: [asterisk-users] chan_motif / res_xmpp problems Reply with quote

Daniel Pocock wrote:
Quote:

I've now replicated my setup on a host with a single IPv4 address and I
am still having trouble with the ICE negotiation.

I am trying to call from Jitsi to Asterisk through a Prosody XMPP
server. Asterisk successfully registers with the XMPP server and
appears to be available in the buddy list in Jitsi. Jitsi is being run
with the "-4" command line option to use IPv4 only just in case Asterisk
doesn't like to see IPv6 ICE candidates.

I try clicking to make an audio-only call from Jitsi. In the Asterisk
logging (xmpp set debug on) I see the incoming "session-initiate" XML
stanza but Asterisk does not send any XML back.

I definitely have "icesupport=yes" in rtp.conf and I've tried it with
and without specifying a TURN server from each end.

Is this working for anybody?

What does your motif.conf configuration file contain? If it is not
configured then it will not be associated with the account and the
Jingle support will not be present.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

--
_____________________________________________________________________
-- 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
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daniel at pocock.pro
Guest





PostPosted: Mon Jul 21, 2014 8:13 am    Post subject: [asterisk-users] chan_motif / res_xmpp problems Reply with quote

On 21/07/14 14:33, Joshua Colp wrote:
Quote:
Daniel Pocock wrote:
Quote:

I've now replicated my setup on a host with a single IPv4 address and I
am still having trouble with the ICE negotiation.

I am trying to call from Jitsi to Asterisk through a Prosody XMPP
server. Asterisk successfully registers with the XMPP server and
appears to be available in the buddy list in Jitsi. Jitsi is being run
with the "-4" command line option to use IPv4 only just in case Asterisk
doesn't like to see IPv6 ICE candidates.

I try clicking to make an audio-only call from Jitsi. In the Asterisk
logging (xmpp set debug on) I see the incoming "session-initiate" XML
stanza but Asterisk does not send any XML back.

I definitely have "icesupport=yes" in rtp.conf and I've tried it with
and without specifying a TURN server from each end.

Is this working for anybody?

What does your motif.conf configuration file contain? If it is not
configured then it will not be associated with the account and the
Jingle support will not be present.


It is largely based on the default config:


[default](!)
disallow=all
allow=ulaw
allow=h264
context=incoming-motif ; Default context that incoming sessions will land in

;maxicecandidates = 10 ; Maximum number of ICE candidates we will offer
;maxpayloads = 30 ; Maximum number of payloads we will offer

[asterisk](default)
disallow=all
allow=alaw
allow=ulaw
transport=ice-udp
connection=asterisk
context=incoming_xmpp



and in xmpp.conf:

[asterisk]
type=client
serverhost=some-host
username=asterisk@some-host
secret=------
usetls=yes
usesasl=yes
status=available
statusmessage="I may be available"
timeout=5


--
_____________________________________________________________________
-- 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

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To UNSUBSCRIBE or update options visit:
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daniel at pocock.pro
Guest





PostPosted: Tue Jul 22, 2014 11:17 am    Post subject: [asterisk-users] chan_motif / res_xmpp problems Reply with quote

On 21/07/14 15:12, Daniel Pocock wrote:
Quote:
On 21/07/14 14:33, Joshua Colp wrote:
Quote:
Daniel Pocock wrote:
Quote:

I've now replicated my setup on a host with a single IPv4 address and I
am still having trouble with the ICE negotiation.

I am trying to call from Jitsi to Asterisk through a Prosody XMPP
server. Asterisk successfully registers with the XMPP server and
appears to be available in the buddy list in Jitsi. Jitsi is being run
with the "-4" command line option to use IPv4 only just in case Asterisk
doesn't like to see IPv6 ICE candidates.

I try clicking to make an audio-only call from Jitsi. In the Asterisk
logging (xmpp set debug on) I see the incoming "session-initiate" XML
stanza but Asterisk does not send any XML back.

I definitely have "icesupport=yes" in rtp.conf and I've tried it with
and without specifying a TURN server from each end.

Is this working for anybody?

What does your motif.conf configuration file contain? If it is not
configured then it will not be associated with the account and the
Jingle support will not be present.


It is largely based on the default config:


[default](!)
disallow=all
allow=ulaw
allow=h264
context=incoming-motif ; Default context that incoming sessions will land in

;maxicecandidates = 10 ; Maximum number of ICE candidates we will offer
;maxpayloads = 30 ; Maximum number of payloads we will offer

[asterisk](default)
disallow=all
allow=alaw
allow=ulaw
transport=ice-udp
connection=asterisk
context=incoming_xmpp



and in xmpp.conf:

[asterisk]
type=client
serverhost=some-host
username=asterisk@some-host
secret=------
usetls=yes
usesasl=yes
status=available
statusmessage="I may be available"
timeout=5




FYI, I'm using the Debian packages, latest is 11.10.2~dfsg-1~bpo70+1

Has the chan_motif / xmpp / ICE stuff changed significantly in 12.x
releases?

Is there any way I can enable ICE debugging?

--
_____________________________________________________________________
-- 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
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jcolp at digium.com
Guest





PostPosted: Tue Jul 22, 2014 11:21 am    Post subject: [asterisk-users] chan_motif / res_xmpp problems Reply with quote

Daniel Pocock wrote:

<snip>

Quote:

FYI, I'm using the Debian packages, latest is 11.10.2~dfsg-1~bpo70+1

Has the chan_motif / xmpp / ICE stuff changed significantly in 12.x
releases?

Nope.

Quote:
Is there any way I can enable ICE debugging?

Not within 11. In 12 there is a module as part of the PJSIP work which
forwards logging messages from the PJ core into Asterisk log messages.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

--
_____________________________________________________________________
-- 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
Back to top
daniel at pocock.pro
Guest





PostPosted: Tue Jul 22, 2014 12:11 pm    Post subject: [asterisk-users] chan_motif / res_xmpp problems Reply with quote

On 22/07/14 18:20, Joshua Colp wrote:
Quote:
Daniel Pocock wrote:

<snip>

Quote:

FYI, I'm using the Debian packages, latest is 11.10.2~dfsg-1~bpo70+1

Has the chan_motif / xmpp / ICE stuff changed significantly in 12.x
releases?

Nope.

Quote:
Is there any way I can enable ICE debugging?

Not within 11. In 12 there is a module as part of the PJSIP work which
forwards logging messages from the PJ core into Asterisk log messages.


Has ice-udp been tested against Jitsi already?

If not, could you please comment on the clients it has been tested with
so I can see if they work against my Asterisk setup?



--
_____________________________________________________________________
-- 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:
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jcolp at digium.com
Guest





PostPosted: Tue Jul 22, 2014 12:14 pm    Post subject: [asterisk-users] chan_motif / res_xmpp problems Reply with quote

Daniel Pocock wrote:
Quote:

On 22/07/14 18:20, Joshua Colp wrote:
Quote:
Daniel Pocock wrote:

<snip>

Quote:
FYI, I'm using the Debian packages, latest is 11.10.2~dfsg-1~bpo70+1

Has the chan_motif / xmpp / ICE stuff changed significantly in 12.x
releases?
Nope.

Quote:
Is there any way I can enable ICE debugging?
Not within 11. In 12 there is a module as part of the PJSIP work which
forwards logging messages from the PJ core into Asterisk log messages.


Has ice-udp been tested against Jitsi already?

It was tested when the code was originally written, since then I don't
know if others have used it. The chan_motif channel driver is primarily
used by people for Google Voice.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

--
_____________________________________________________________________
-- 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
Back to top
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