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
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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-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 > <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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150722/d3cc6422/attachment-0001.html>
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<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 Andres
Sent: Wednesday, 22 July 2015 9:29 PM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Cisco 7940 and PJSIP registration
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-bounces at
lists.digium.com<mailto: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
--
Technical Support
http://www.cellroute.net
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