VoIP Mailing List Archives

[bord at staff.onthenet...]

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;

<![if !supportLists]>- <![endif]>Asterisk 13.4 with FreePBX 12.
<![if !supportLists]>- <![endif]>Migrating from Asterisk 11 / FreePBX 2.11
<![if !supportLists]>- <![endif]>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@172.22.4.8>;tag=001469a7180c0011603d4433-6cef1ff3.
To: <sip:233@172.22.4.8>.
Call-ID: 001469a7-180c0002-58faebd6-05b99917@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@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.....@................&..REGISTER sip:172.22.4.8 SIP/2.0.
Via: SIP/2.0/UDP 172.22.3.228:5060;branch=z9hG4bK505b3494.
From: <sip:233@172.22.4.8>;tag=001469a7180c0011603d4433-6cef1ff3.
To: <sip:233@172.22.4.8>.
Call-ID: 001469a7-180c0002-58faebd6-05b99917@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@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. – 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)
www.OntheNet.com.au

[me at nileshgr.com]

 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@staff.onthenet.com.au (bord@staff.onthenet.com.au)> wrote:

[bord at staff.onthenet...]

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)
www.OntheNet.com.au

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@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@staff.onthenet.com.au (bord@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 [url=sip:172.22.4.8]sip:172.22.4.8[/url] SIP/2.0.
Via: SIP/2.0/UDP 172.22.3.228:5060;branch=z9hG4bK505b3494.
From: <sip:233@172.22.4.8 ([email]sip%3A233@172.22.4.8[/email])>;tag=001469a7180c0011603d4433-6cef1ff3.
To: <sip:233@172.22.4.8 ([email]sip%3A233@172.22.4.8[/email])>.
Call-ID: 001469a7-180c0002-58faebd6-05b99917@172.22.3.228 (001469a7-180c0002-58faebd6-05b99917@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: <[url=sip:233@172.22.3.228:5060;user=phone;transport=udp]sip:233@172.22.3.228:5060;user=phone;transport=udp[/url]>;+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.....@................&..REGISTER ([email]...@................&..REGISTER[/email]) [url=sip:172.22.4.8]sip:172.22.4.8[/url] SIP/2.0.
Via: SIP/2.0/UDP 172.22.3.228:5060;branch=z9hG4bK505b3494.
From: <sip:233@172.22.4.8 ([email]sip%3A233@172.22.4.8[/email])>;tag=001469a7180c0011603d4433-6cef1ff3.
To: <sip:233@172.22.4.8 ([email]sip%3A233@172.22.4.8[/email])>.
Call-ID: 001469a7-180c0002-58faebd6-05b99917@172.22.3.228 (001469a7-180c0002-58faebd6-05b99917@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: <[url=sip:233@172.22.3.228:5060;user=phone;transport=udp]sip:233@172.22.3.228:5060;user=phone;transport=udp[/url]>;+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. – 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)
www.OntheNet.com.au




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[andres at telesip.net]

On 7/22/15 1:38 AM, Brendan Ord wrote:

[bord at staff.onthenet...]

Thank you.

I read that last yesterday afternoon, and I could’ve sworn I tried that but I will look into it again (I’ve tried so many different things it was getting cloudy what I’ve tried and what worked etc, combined that the extension config gets messed up after playing with it so much so I’m often recreating it as well).

I also found a bug report in the FreePBX bug tracker http://issues.freepbx.org/browse/FREEPBX-8517

It’s not exactly the same, but it’s very similar and the closing comment was “limitation of pjsip”. I might be getting ahead of myself, but would anyone be able to comment on that? Anyway, this looks like a FreePBX issue so this isn’t the ideal forum to discuss their bugs.


Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au


From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andres
Sent: Wednesday, 22 July 2015 9:29 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco 7940 and PJSIP registration



On 7/22/15 1:38 AM, Brendan Ord wrote:

[bord at staff.onthenet...]

Hello,

Putting this into pjsip.endpoint_custom.conf;

[233](+)

force_rport=no

Didn’t work – as can be seen below, Asterisk is still replying on the source port instead of 5060;

#
U 172.22.4.50:51037 -> 172.22.4.8:5060
REGISTER [url=sip:172.22.4.8]sip:172.22.4.8[/url] SIP/2.0.
Via: SIP/2.0/UDP 172.22.4.50:5060;branch=z9hG4bK6c6b20ee.
From: <[url=sip:233@172.22.4.8]sip:233@172.22.4.8[/url]>;tag=001469a7180c00a6368fbc8e-7fedf772.
To: <[url=sip:233@172.22.4.8]sip:233@172.22.4.8[/url]>.
Call-ID: 001469a7-180c0002-58f81e51-5a17e8fb@172.22.4.50 (001469a7-180c0002-58f81e51-5a17e8fb@172.22.4.50).
Max-Forwards: 70.
Date: Thu, 23 Jul 2015 02:53:26 GMT.
CSeq: 265 REGISTER.
User-Agent: Cisco-CP7940G/8.0.
Contact: <[url=sip:233@172.22.4.50:5060;user=phone;transport=udp]sip:233@172.22.4.50:5060;user=phone;transport=udp[/url]>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001469a7180c>";+u.sip!model.ccm.cisco.com="8".
Content-Length: 0.
Expires: 120.
.

#
U 172.22.4.8:5060 -> 172.22.4.50:51037
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 172.22.4.50:5060;rport=51037;received=172.22.4.50;branch=z9hG4bK6c6b20ee.
Call-ID: 001469a7-180c0002-58f81e51-5a17e8fb@172.22.4.50 (001469a7-180c0002-58f81e51-5a17e8fb@172.22.4.50).
From: <[url=sip:233@172.22.4.8]sip:233@172.22.4.8[/url]>;tag=001469a7180c00a6368fbc8e-7fedf772.
To: <[url=sip:233@172.22.4.8]sip:233@172.22.4.8[/url]>;tag=z9hG4bK6c6b20ee.
CSeq: 265 REGISTER.
WWW-Authenticate: Digest realm="asterisk",nonce="1437620030/5d32272e81266a723b2c090074799fe2",opaque="65e177e35509170b",algorithm=md5,qop="auth".
Server: FPBX-AsteriskNOW-12.0.73(13.4.0).
Content-Length: 0.
.


Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au


From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brendan Ord
Sent: Thursday, 23 July 2015 10:51 AM
To: 'andres@telesip.net'; asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco 7940 and PJSIP registration



Thank you.

I read that last yesterday afternoon, and I could’ve sworn I tried that but I will look into it again (I’ve tried so many different things it was getting cloudy what I’ve tried and what worked etc, combined that the extension config gets messed up after playing with it so much so I’m often recreating it as well).

I also found a bug report in the FreePBX bug tracker http://issues.freepbx.org/browse/FREEPBX-8517

It’s not exactly the same, but it’s very similar and the closing comment was “limitation of pjsip”. I might be getting ahead of myself, but would anyone be able to comment on that? Anyway, this looks like a FreePBX issue so this isn’t the ideal forum to discuss their bugs.


Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au


From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Andres
Sent: Wednesday, 22 July 2015 9:29 PM
To: asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)
Subject: Re: [asterisk-users] Cisco 7940 and PJSIP registration



On 7/22/15 1:38 AM, Brendan Ord wrote:

[bord at staff.onthenet...]

Another FreePBX forum member has logged a bug report for me, so this thread can be closed.

http://issues.freepbx.org/browse/FREEPBX-9810

Thanks for everyone’s input with this problem.

Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au


From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brendan Ord
Sent: Thursday, 23 July 2015 1:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Cisco 7940 and PJSIP registration



Hello,

Putting this into pjsip.endpoint_custom.conf;

[233](+)

force_rport=no

Didn’t work – as can be seen below, Asterisk is still replying on the source port instead of 5060;

#
U 172.22.4.50:51037 -> 172.22.4.8:5060
REGISTER [url=sip:172.22.4.8]sip:172.22.4.8[/url] SIP/2.0.
Via: SIP/2.0/UDP 172.22.4.50:5060;branch=z9hG4bK6c6b20ee.
From: <[url=sip:233@172.22.4.8]sip:233@172.22.4.8[/url]>;tag=001469a7180c00a6368fbc8e-7fedf772.
To: <[url=sip:233@172.22.4.8]sip:233@172.22.4.8[/url]>.
Call-ID: 001469a7-180c0002-58f81e51-5a17e8fb@172.22.4.50 (001469a7-180c0002-58f81e51-5a17e8fb@172.22.4.50).
Max-Forwards: 70.
Date: Thu, 23 Jul 2015 02:53:26 GMT.
CSeq: 265 REGISTER.
User-Agent: Cisco-CP7940G/8.0.
Contact: <[url=sip:233@172.22.4.50:5060;user=phone;transport=udp]sip:233@172.22.4.50:5060;user=phone;transport=udp[/url]>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-001469a7180c>";+u.sip!model.ccm.cisco.com="8".
Content-Length: 0.
Expires: 120.
.

#
U 172.22.4.8:5060 -> 172.22.4.50:51037
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 172.22.4.50:5060;rport=51037;received=172.22.4.50;branch=z9hG4bK6c6b20ee.
Call-ID: 001469a7-180c0002-58f81e51-5a17e8fb@172.22.4.50 (001469a7-180c0002-58f81e51-5a17e8fb@172.22.4.50).
From: <[url=sip:233@172.22.4.8]sip:233@172.22.4.8[/url]>;tag=001469a7180c00a6368fbc8e-7fedf772.
To: <[url=sip:233@172.22.4.8]sip:233@172.22.4.8[/url]>;tag=z9hG4bK6c6b20ee.
CSeq: 265 REGISTER.
WWW-Authenticate: Digest realm="asterisk",nonce="1437620030/5d32272e81266a723b2c090074799fe2",opaque="65e177e35509170b",algorithm=md5,qop="auth".
Server: FPBX-AsteriskNOW-12.0.73(13.4.0).
Content-Length: 0.
.


Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au


From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Brendan Ord
Sent: Thursday, 23 July 2015 10:51 AM
To: 'andres@telesip.net'; asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)
Subject: Re: [asterisk-users] Cisco 7940 and PJSIP registration



Thank you.

I read that last yesterday afternoon, and I could’ve sworn I tried that but I will look into it again (I’ve tried so many different things it was getting cloudy what I’ve tried and what worked etc, combined that the extension config gets messed up after playing with it so much so I’m often recreating it as well).

I also found a bug report in the FreePBX bug tracker http://issues.freepbx.org/browse/FREEPBX-8517

It’s not exactly the same, but it’s very similar and the closing comment was “limitation of pjsip”. I might be getting ahead of myself, but would anyone be able to comment on that? Anyway, this looks like a FreePBX issue so this isn’t the ideal forum to discuss their bugs.


Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map)
www.OntheNet.com.au


From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Andres
Sent: Wednesday, 22 July 2015 9:29 PM
To: asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)
Subject: Re: [asterisk-users] Cisco 7940 and PJSIP registration



On 7/22/15 1:38 AM, Brendan Ord wrote: