[Home]

Summary:ASTERISK-26248: chan_pjsip: Error when calling PJSIP client with domain specified
Reporter:Norbert Varga (codwell)Labels:
Date Opened:2016-07-28 05:12:07Date Closed:2017-02-13 09:38:19.000-0600
Priority:MinorRegression?
Status:Closed/CompleteComponents:Channels/chan_pjsip Resources/res_pjsip
Versions:13.9.1 Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) multidomain-pjsip-patch.diff
Description:I'd like to use endpoint with different domains in my system. I have the following config in pjsip.conf:
{code}
[1000@mypbx.com]
type = endpoint
transport = transport-udp-nat
disallow = all
allow = ulaw
direct_media = no
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
context = test
auth = 1000
aors = 1000

[1000]
type = auth
auth_type = userpass
username = test
password = xy

[1000]
type = aor
max_contacts = 1
remove_existing = yes
{code}

In extensions.conf I have:
{code}
[test]
exten => _ZXXX,1,Dial(PJSIP/${EXTEN}@mypbx.com)
{code}

Register works. When I try to dial 1000 from another extensions, I get this:
{code}
ERROR[38010]: chan_pjsip.c:1907 request: Unable to create PJSIP channel - endpoint 'mypbx.com' was not found
WARNING[38308][C-00000002]: app_dial.c:2429 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
{code}

The problem is that in the dialplan I have to specify the full name of the endpoint (1000@mypbx.com), but in chan_pjsip above line 1907 the host part is taken as endpoint. The only workaround which is worked, but it looks ugly, and I had to look in the code how to workaround this:
{code}
exten => _ZXXX,1,Dial(PJSIP/@${EXTEN}@mypbx.com)
{code}
Comments:By: Asterisk Team (asteriskteam) 2016-07-28 05:12:07.840-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].
By: Norbert Varga (codwell) 2016-07-31 07:06:19.269-0500

I recently upgraded to 13.10.0. Now it seems calling a PJSIP client is not working, either for multi-domain and single domain account, I got the same error. This configuration was working in 13.9.1. Could you please suggest what's the problem here?
{code}
   -- Executing [1001@pbxtest:1] Dial("PJSIP/1000-00000000", "PJSIP/1001") in new stack
[Jul 31 13:45:22] ERROR[5761]: res_pjsip.c:2785 ast_sip_create_dialog_uac: Could not create dialog to endpoint '1001' as URI '1001' is not valid
[Jul 31 13:45:22] ERROR[5761]: chan_pjsip.c:1921 request: Failed to create outgoing session to endpoint '1001'
[Jul 31 13:45:22] WARNING[5819][C-00000000]: app_dial.c:2432 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
{code}

pjsip.conf:
{code}
[1000]
type = endpoint
transport = transport-udp-nat
disallow = all
allow = ulaw
direct_media = no
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
context = pbxtest
auth = pbxtest-1000
aors = 1000

[pbxtest-1000]
type = auth
auth_type = userpass
username = 1000
password = xyz

[1000]
type = aor
max_contacts = 1
remove_existing = yes

[1001]
type = endpoint
transport = transport-udp-nat
disallow = all
allow = ulaw
direct_media = no
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
context = pbxtest
auth = pbxtest-1001
aors = 1001

[pbxtest-1001]
type = auth
auth_type = userpass
username = 1001
password = xyz

[1001]
type = aor
max_contacts = 1
remove_existing = yes
{code}

extensions.conf:
{code}
[pbxtest]
exten => _ZXXX,1,Dial(PJSIP/${EXTEN})
{code}
By: Joshua C. Colp (jcolp) 2016-07-31 07:09:19.055-0500

Are the devices registered? If not then you will get the message you see.
By: Norbert Varga (codwell) 2016-07-31 07:29:41.054-0500

Thank you, it's working! That was the problem. Sorry about the noise.
By: Norbert Varga (codwell) 2016-08-01 05:10:41.481-0500

The original problem still persist, and currently I'm using call queues and I couldn't make a workaround. I have the following call queue:
{code}
[pbxtest-sales]
member => PJSIP/1000@mypbx.com
member => PJSIP/1001@mypbx.com
{code}

Queue show command shows the members correctly. Extensions.conf:
{code}
[pbxtest]
exten => 2000,1,Answer()
same => n,Queue(pbxtest-sales)
same => n,Hangup()

; exten => 1000,1,Dial(PJSIP/@1000@mypbx.com)
exten => 1000,1,Dial(PJSIP/1000@mypbx.com)
; exten => 1001,1,Dial(PJSIP/@1001@mypbx.com)
exten => 1001,1,Dial(PJSIP/1001@mypbx.com)
{code}

The result is:
{code}
Executing [2000@pbxtest:1] Answer("PJSIP/1000@mypbx.com-00000025", "") in new stack
      > 0x7ff0e8009470 -- Probation passed - setting RTP source address to 92.249.245.153:4012
   -- Executing [2000@pbxtest:2] Queue("PJSIP/1000@mypbx.com-00000025", "pbxtest-sales") in new stack
   -- Started music on hold, class 'default', on channel 'PJSIP/1000@mypbx.com-00000025'
[2016-08-01 11:59:47] ERROR[7165]: chan_pjsip.c:1915 request: Unable to create PJSIP channel - endpoint 'mypbx.com' was not found
{code}

I tried both versions in extension.conf (commented out). I think the reason behind this when the members are called, it's called directly bypassing routing. So it seems the original problem in chan_pjsip.c should be fixed...
By: Norbert Varga (codwell) 2017-01-10 10:41:53.112-0600

See my patch for fix this issue.
By: Rusty Newton (rnewton) 2017-01-10 12:44:42.856-0600

[~codwell] go ahead and get the patch up on Gerrit. Here is the overall process for reference: https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process

Here is the Gerrit specific bit: https://wiki.asterisk.org/wiki/display/AST/Gerrit+Usage
By: Friendly Automation (friendly-automation) 2017-02-13 09:38:20.150-0600

Change 4719 merged by zuul:
chan_pjsip: Multidomain endpoint finding on call

[https://gerrit.asterisk.org/4719|https://gerrit.asterisk.org/4719]
By: Friendly Automation (friendly-automation) 2017-02-13 09:44:46.146-0600

Change 4870 merged by zuul:
chan_pjsip: Multidomain endpoint finding on call

[https://gerrit.asterisk.org/4870|https://gerrit.asterisk.org/4870]
By: Friendly Automation (friendly-automation) 2017-02-13 09:44:48.749-0600

Change 4869 merged by zuul:
chan_pjsip: Multidomain endpoint finding on call

[https://gerrit.asterisk.org/4869|https://gerrit.asterisk.org/4869]