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Brendan Ord

2015-Jul-22 01:37 UTC

[asterisk-users] Cisco 7940 and PJSIP registration

Hi list,

I've been googling this issue and found some good resources however I am
still running into problems with the following combo ... Here's my story;


-      Asterisk 13.4 with FreePBX 12.

-      Migrating from Asterisk 11 / FreePBX 2.11

-      Mix of Cisco 79xx handsets, mostly 7940G's.

My problems started with (the very common) issue of the 7940 not replying to 401
UNAUTHORIZED with a second REGISTER containing the auth digest details.  A quick
Google found a heap of information in various forums, all with replies from
Joshua Colp stating that force_rport=no needs to be set for these endpoints, see
http://forums.digium.com/viewtopic.php?f=1&t=91699

So, (being that this is FreePBX and the main conf files are controlled by that)
I jumped into /etc/asterisk/pjsip.endpoint_custom.conf and added;

[233]
force_rport=no

Reloaded everything, recreated the extension and tested again, watching what
goes between this endpoint with 'ngrep -W byline host 172.22.3.228' and
now I get something which I don't fully understand;

U 172.22.3.228:51440 -> 172.22.4.8:5060
REGISTER sip:172.22.4.8 SIP/2.0.
Via: SIP/2.0/UDP 172.22.3.228:5060;branch=z9hG4bK505b3494.
From: <sip:233 at 172.22.4.8>;tag=001469a7180c0011603d4433-6cef1ff3.
To: <sip:233 at 172.22.4.8>.
Call-ID: 001469a7-180c0002-58faebd6-05b99917 at 172.22.3.228.
Max-Forwards: 70.
Date: Wed, 22 Jul 2015 00:41:48 GMT.
CSeq: 114 REGISTER.
User-Agent: Cisco-CP7940G/8.0.
Contact: <sip:233 at
172.22.3.228:5060;user=phone;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001469a7180c>";+u.sip!model.ccm.cisco.com="8".
Content-Length: 0.
Expires: 120.
.

#
I 172.22.4.8 -> 172.22.3.228 3:3
....E..:)... at ................&..REGISTER sip:172.22.4.8 SIP/2.0.
Via: SIP/2.0/UDP 172.22.3.228:5060;branch=z9hG4bK505b3494.
From: <sip:233 at 172.22.4.8>;tag=001469a7180c0011603d4433-6cef1ff3.
To: <sip:233 at 172.22.4.8>.
Call-ID: 001469a7-180c0002-58faebd6-05b99917 at 172.22.3.228.
Max-Forwards: 70.
Date: Wed, 22 Jul 2015 00:41:48 GMT.
CSeq: 114 REGISTER.
User-Agent: Cisco-CP7940G/8.0.
Contact: <sip:233 at
172.22.3.228:5060;user=phone;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001469a7180c>";+u.sip!model.ccm.cisco.com="8".
Content-Lengt

I don't understand this reply from Asterisk (172.22.4.8) - why it's not
complete and what's this 3:3?

If anyone has input or experience with this problem I would be forever grateful.
I have read that people can get these handsets working with chan_sip (and,
indeed they do, as these handsets are working perfectly using chan_sip in
Asterisk 11), but I would really like to keep everything using pjsip (for the
reason that, this is where development and improvements are heading, and I like
to be using the best technology if possible).

Thank you...

Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227
(Map<https://goo.gl/maps/p25WF>)
www.OntheNet.com.au<http://www.onthenet.com.au/>

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Nilesh Govindrajan

2015-Jul-22 01:45 UTC

head link

[asterisk-users] Cisco 7940 and PJSIP registration

I had exact same issue with pjsip instead of sip - I was able to solve it
by setting the password to blank. But I switched to asterisk 11 because the
chan_mobile module was giving me troubles in 13.

On Wed, Jul 22, 2015 at 7:07 AM, Brendan Ord <bord at
staff.onthenet.com.au>
wrote:
>  Hi list,
>
>
>
> I?ve been googling this issue and found some good resources however I am
> still running into problems with the following combo ? Here?s my story;
>
>
>
> -      Asterisk 13.4 with FreePBX 12.
>
> -      Migrating from Asterisk 11 / FreePBX 2.11
>
> -      Mix of Cisco 79xx handsets, mostly 7940G?s.
>
>
>
> My problems started with (the very common) issue of the 7940 not replying
> to 401 UNAUTHORIZED with a second REGISTER containing the auth digest
> details.  A quick Google found a heap of information in various forums, all
> with replies from Joshua Colp stating that force_rport=no needs to be set
> for these endpoints, see
> http://forums.digium.com/viewtopic.php?f=1&t=91699
>
>
>
> So, (being that this is FreePBX and the main conf files are controlled by
> that) I jumped into /etc/asterisk/pjsip.endpoint_custom.conf and added;
>
>
>
> [233]
>
> force_rport=no
>
>
>
> Reloaded everything, recreated the extension and tested again, watching
> what goes between this endpoint with ?ngrep ?W byline host 172.22.3.228?
> and now I get something which I don?t fully understand;
>
>
>
> U 172.22.3.228:51440 -> 172.22.4.8:5060
>
> REGISTER sip:172.22.4.8 SIP/2.0.
>
> Via: SIP/2.0/UDP 172.22.3.228:5060;branch=z9hG4bK505b3494.
>
> From: <sip:233 at 172.22.4.8>;tag=001469a7180c0011603d4433-6cef1ff3.
>
> To: <sip:233 at 172.22.4.8>.
>
> Call-ID: 001469a7-180c0002-58faebd6-05b99917 at 172.22.3.228.
>
> Max-Forwards: 70.
>
> Date: Wed, 22 Jul 2015 00:41:48 GMT.
>
> CSeq: 114 REGISTER.
>
> User-Agent: Cisco-CP7940G/8.0.
>
> Contact: <sip:233 at 172.22.3.228:5060
>
;user=phone;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001469a7180c>";+u.sip!
> model.ccm.cisco.com="8".
>
> Content-Length: 0.
>
> Expires: 120.
>
> .
>
>
>
> #
>
> I 172.22.4.8 -> 172.22.3.228 3:3
>
> ....E..:)... at ................&..REGISTER sip:172.22.4.8 SIP/2.0.
>
> Via: SIP/2.0/UDP 172.22.3.228:5060;branch=z9hG4bK505b3494.
>
> From: <sip:233 at 172.22.4.8>;tag=001469a7180c0011603d4433-6cef1ff3.
>
> To: <sip:233 at 172.22.4.8>.
>
> Call-ID: 001469a7-180c0002-58faebd6-05b99917 at 172.22.3.228.
>
> Max-Forwards: 70.
>
> Date: Wed, 22 Jul 2015 00:41:48 GMT.
>
> CSeq: 114 REGISTER.
>
> User-Agent: Cisco-CP7940G/8.0.
>
> Contact: <sip:233 at 172.22.3.228:5060
>
;user=phone;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001469a7180c>";+u.sip!
> model.ccm.cisco.com="8".
>
> Content-Lengt
>
>
>
> I don?t understand this reply from Asterisk (172.22.4.8) ? why it?s not
> complete and what?s this 3:3?
>
>
>
> If anyone has input or experience with this problem I would be forever
> grateful.  I have read that people can get these handsets working with
> chan_sip (and, indeed they do, as these handsets are working perfectly
> using chan_sip in Asterisk 11), but I would really like to keep everything
> using pjsip (for the reason that, this is where development and
> improvements are heading, and I like to be using the best technology if
> possible).
>
>
>
> Thank you?
>
>
>
> Brendan Ord
> OntheNet - Network Engineer
> P 07 5553 9222
> F 07 5593 3557
> Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map
> <https://goo.gl/maps/p25WF>)
> www.OntheNet.com.au <http://www.onthenet.com.au/>
>
>
>
> --
> _____________________________________________________________________
> -- 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
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Brendan Ord

2015-Jul-22 05:38 UTC

head link

[asterisk-users] Cisco 7940 and PJSIP registration

I?ve gotten to the bottom of this;

Seems that the pjsip.endpoint_custom.conf isn?t getting included properly, or my
syntax is wrong.

If I put force_rport=no into pjsip.endpoint.conf and reload only Asterisk,
everything works perfectly.  Unfortunately, I?m using FreePBX, so it owns this
file and my changes won?t persist a FreePBX reload.

Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227
(Map<https://goo.gl/maps/p25WF>)
www.OntheNet.com.au<http://www.onthenet.com.au/>

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Nilesh Govindrajan
Sent: Wednesday, 22 July 2015 11:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7940 and PJSIP registration

 I had exact same issue with pjsip instead of sip - I was able to solve it by
setting the password to blank. But I switched to asterisk 11 because the
chan_mobile module was giving me troubles in 13.

On Wed, Jul 22, 2015 at 7:07 AM, Brendan Ord <bord at
staff.onthenet.com.au<mailto:bord at staff.onthenet.com.au>> wrote:
Hi list,

I?ve been googling this issue and found some good resources however I am still
running into problems with the following combo ? Here?s my story;


-      Asterisk 13.4 with FreePBX 12.

-      Migrating from Asterisk 11 / FreePBX 2.11

-      Mix of Cisco 79xx handsets, mostly 7940G?s.

My problems started with (the very common) issue of the 7940 not replying to 401
UNAUTHORIZED with a second REGISTER containing the auth digest details.  A quick
Google found a heap of information in various forums, all with replies from
Joshua Colp stating that force_rport=no needs to be set for these endpoints, see
http://forums.digium.com/viewtopic.php?f=1&t=91699

So, (being that this is FreePBX and the main conf files are controlled by that)
I jumped into /etc/asterisk/pjsip.endpoint_custom.conf and added;

[233]
force_rport=no

Reloaded everything, recreated the extension and tested again, watching what
goes between this endpoint with ?ngrep ?W byline host 172.22.3.228? and now I
get something which I don?t fully understand;

U 172.22.3.228:51440<http://172.22.3.228:51440> ->
172.22.4.8:5060<http://172.22.4.8:5060>
REGISTER sip:172.22.4.8 SIP/2.0.
Via: SIP/2.0/UDP 172.22.3.228:5060;branch=z9hG4bK505b3494.
From: <sip:233 at 172.22.4.8<mailto:sip%3A233 at
172.22.4.8>>;tag=001469a7180c0011603d4433-6cef1ff3.
To: <sip:233 at 172.22.4.8<mailto:sip%3A233 at 172.22.4.8>>.
Call-ID: 001469a7-180c0002-58faebd6-05b99917 at
172.22.3.228<mailto:001469a7-180c0002-58faebd6-05b99917 at 172.22.3.228>.
Max-Forwards: 70.
Date: Wed, 22 Jul 2015 00:41:48 GMT.
CSeq: 114 REGISTER.
User-Agent: Cisco-CP7940G/8.0.
Contact: <sip:233 at
172.22.3.228:5060;user=phone;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001469a7180c>";+u.sip!model.ccm.cisco.com<http://model.ccm.cisco.com>="8".
Content-Length: 0.
Expires: 120.
.

#
I 172.22.4.8 -> 172.22.3.228 3:3
....E..:)... at ................&..REGISTER<mailto:... at
................&..REGISTER> sip:172.22.4.8 SIP/2.0.
Via: SIP/2.0/UDP 172.22.3.228:5060;branch=z9hG4bK505b3494.
From: <sip:233 at 172.22.4.8<mailto:sip%3A233 at
172.22.4.8>>;tag=001469a7180c0011603d4433-6cef1ff3.
To: <sip:233 at 172.22.4.8<mailto:sip%3A233 at 172.22.4.8>>.
Call-ID: 001469a7-180c0002-58faebd6-05b99917 at
172.22.3.228<mailto:001469a7-180c0002-58faebd6-05b99917 at 172.22.3.228>.
Max-Forwards: 70.
Date: Wed, 22 Jul 2015 00:41:48 GMT.
CSeq: 114 REGISTER.
User-Agent: Cisco-CP7940G/8.0.
Contact: <sip:233 at
172.22.3.228:5060;user=phone;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001469a7180c>";+u.sip!model.ccm.cisco.com<http://model.ccm.cisco.com>="8".
Content-Lengt

I don?t understand this reply from Asterisk (172.22.4.8) ? why it?s not complete
and what?s this 3:3?

If anyone has input or experience with this problem I would be forever grateful.
I have read that people can get these handsets working with chan_sip (and,
indeed they do, as these handsets are working perfectly using chan_sip in
Asterisk 11), but I would really like to keep everything using pjsip (for the
reason that, this is where development and improvements are heading, and I like
to be using the best technology if possible).

Thank you?

Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227
(Map<https://goo.gl/maps/p25WF>)
www.OntheNet.com.au<http://www.onthenet.com.au/>


--
_____________________________________________________________________
-- 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

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