On 7/22/15 1:38 AM, Brendan Ord wrote:
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.
Last time I checked you have to put a plus sign to combine parameters
from main and custom file. Like this:
[233](+)
force_rport=no
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-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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
<b...@staff.onthenet.com.au <mailto:b...@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@172.22.4.8
<mailto:sip%3A233@172.22.4.8>>;tag=001469a7180c0011603d4433-6cef1ff3.
To: <sip:233@172.22.4.8 <mailto:sip%3A233@172.22.4.8>>.
Call-ID: 001469a7-180c0002-58faebd6-05b99917@172.22.3.228
<mailto: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
<http://model.ccm.cisco.com>="8".
Content-Length: 0.
Expires: 120.
.
#
I 172.22.4.8 -> 172.22.3.228 3:3
....E..:)...@................&..REGISTER
<mailto:...@................&..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
<mailto:sip%3A233@172.22.4.8>>;tag=001469a7180c0011603d4433-6cef1ff3.
To: <sip:233@172.22.4.8 <mailto:sip%3A233@172.22.4.8>>.
Call-ID: 001469a7-180c0002-58faebd6-05b99917@172.22.3.228
<mailto: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
<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
<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
--
Technical Support
http://www.cellroute.net
--
_____________________________________________________________________
-- 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
