Ip register in pjsip

Asterisk Asterisk General
22

Hi jcolp :wink:

ok now in my ip phones appears registered but in cli Status appears Unknown

look :

pjsip show contacts

Contact: <Aor/ContactUri…> <Status…> <RTT(ms)…>

Contact: 20021/sip:20021@192.168.0.20:5060 Unknown nan
Contact: 20021/sip:20021@192.168.0.20:5070 Unknown nan

and when i make command , pjsip show registrations
No objects found.

how i can fixed ?

Regards

23

Hi ,

how i can fstop these warning ?

Asterisk 13.2.0, Copyright © 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 13.2.0 currently running on ipbx (pid = 22583)
[Jun 14 22:44:17] WARNING[26895]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event as-feature-event
[Jun 14 22:44:19] WARNING[30367]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event as-feature-event
[Jun 14 22:44:25] WARNING[2928]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event as-feature-event
[Jun 14 22:44:32] WARNING[26895]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event as-feature-event
ipbx*CLI>

Regards

24

Disable the subscription request, for the unsupported feature, in the peer.

This might help: http://community.polycom.com/t5/VoIP/SUBSCRIBE-for-as-feature-event/td-p/23606

25

Hi

these appears on pjsip

[Jun 15 18:10:21] WARNING[31571]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event as-feature-event
where i need fixed these i phons yealink or in asterisk ??

Regards,

26

The phone, unless you want to write code to handle subscriptions to that event type.

27

Hi
No , the only thing I want, and this stop.

28

The answer from the original thread and from @david551 is correct. You either need to adjust logging such that it’s not output, remove the message from the source code, or stop the device from trying to subscribe.

29

HI,

Ok , How ,where ?

30

The logger.conf file can be used to configure logging, as for removing the message you’d have to find it in the source code, delete it, and rebuild Asterisk.

31

For the device, you will need to consult the documentation for the device.

32

Hi ,

i´m using yelling phones T46 and T42 .

33

You should try their forum for configuration help on their phones.

http://forum.yealink.com/forum/

34

Hi ok tanks for your replay ,

i thing all ready fixed the problem of event .

i try make a call betwenn to endpoints but no sound .

<— Received SIP request (445 bytes) from UDP:192.168.0.14:5062 —>
BYE sip:0af296c9-e9da-480b-a66a-b4d46e833863@192.168.0.103:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.14:5062;branch=z9hG4bK3829646500
From: sip:20050@192.168.0.14;tag=1281203426
To: “Arm_1” sip:20051@192.168.0.103;tag=34944694-8c21-4edd-b9af-79dd4bce3ff5
Call-ID: 307e9e67-4d07-4308-a9de-da0adf3e38d9
CSeq: 31897 BYE
Contact: sip:20050@192.168.0.14:5062
Max-Forwards: 70
User-Agent: Yealink SIP-W52P 25.73.0.40
Content-Length: 0

<— Transmitting SIP response (323 bytes) to UDP:192.168.0.14:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.14:5062;rport=5062;received=192.168.0.14;branch=z9hG4bK3829646500
Call-ID: 307e9e67-4d07-4308-a9de-da0adf3e38d9
From: sip:20050@192.168.0.14;tag=1281203426
To: “Arm_1” sip:20051@192.168.0.103;tag=34944694-8c21-4edd-b9af-79dd4bce3ff5
CSeq: 31897 BYE
Content-Length: 0

-- Channel PJSIP/20050-00000005 left 'native_rtp' basic-bridge <6a988407-b338-4579-867b-b74fdb5af181>
-- Channel PJSIP/20051-00000004 left 'native_rtp' basic-bridge <6a988407-b338-4579-867b-b74fdb5af181>

== Spawn extension (default, 20050, 2) exited non-zero on ‘PJSIP/20051-00000004’
<— Transmitting SIP request (359 bytes) to UDP:192.168.0.14:5063 —>
BYE sip:20051@192.168.0.14:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;rport;branch=z9hG4bKPj843e02bd-f7a6-49ee-8869-0759ff2770e8
From: sip:20050@192.168.0.29;tag=a9b2a58b-8764-4be1-b1f6-e78a542fbb3d
To: “Arm_1” sip:20051@192.168.0.29;tag=807932943
Call-ID: 3547949677@192.168.0.14
CSeq: 32367 BYE
Reason: Q.850;cause=16
Content-Length: 0

<— Received SIP response (350 bytes) from UDP:192.168.0.14:5063 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.103:5060;rport;branch=z9hG4bKPj843e02bd-f7a6-49ee-8869-0759ff2770e8
From: sip:20050@192.168.0.29;tag=a9b2a58b-8764-4be1-b1f6-e78a542fbb3d
To: “Arm_1” sip:20051@192.168.0.29;tag=807932943
Call-ID: 3547949677@192.168.0.14
CSeq: 32367 BYE
User-Agent: Yealink SIP-W52P 25.73.0.40
Content-Length: 0

Regards,