PJSIP - Outbound Registration

@Deathlok92 wrote:

Hi all,

I'm having a problem with a trunk registration.

Actually my freePBX is 100% PJSIP (from trunk to extensions) I have 4 provider trunks, 3 of them work perfectly.

I'm not able to make the last one get registered.

I have some logs, but I'm pretty new to asterisk, can you please me to troubleshooting the issue?

This are logs of the trunk registering

[2017-06-30 22:43:43] VERBOSE[2783] res_pjsip_logger.c: <--- Transmitting SIP request (762 bytes) to UDP:217.19.154.186:5060 --->
REGISTER sip:tree. fiware. com:5060 SIP/2.0
Via: SIP/2.0/UDP 217.19.146.213:5060;rport;branch=z9hG4bKPjae33dd33-aa4a-4757-b948-8ff2825da362
From: <sip:390144485377@tree. fiware. com>;tag=fb7e5cf3-5367-458c-a05b-55a477aca897
To: <sip:390144485377@tree. fiware. com>
Call-ID: 5ebbfc80-b8bb-41b8-a2cb-b6f46e16b1fd
CSeq: 40299 REGISTER
Contact: <sip:s@217.19.146.213:5060;line=zmprapy>
Expires: 60
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Max-Forwards: 70
User-Agent: FPBX-14.0.1rc1.30(13.16.0)
Authorization: Digest username="390144485377", realm="asterisk", nonce="1d415d08", uri="sip:tree. fiware. com:5060", response="806e3ca20292e5f6fa3bf3053e90ae95", algorithm=MD5
Content-Length: 0


[2017-06-30 22:43:43] VERBOSE[2782] res_pjsip_logger.c: <--- Received SIP request (575 bytes) from UDP:217.19.154.186:5060 --->
OPTIONS sip:s@217.19.146.213:5060;line=zmprapy SIP/2.0
Via: SIP/2.0/UDP 217.19.154.186:5060;branch=z9hG4bK2684817d;rport
Max-Forwards: 70
From: "asterisk" <sip:peer@217.19.154.186>;tag=as310eb014
To: <sip:s@217.19.146.213:5060;line=zmprapy>
Contact: <sip:peer@217.19.154.186:5060>
Call-ID: 6fbb262504eebe5140d3cfe41fe1bf4d@217.19.154.186:5060
CSeq: 102 OPTIONS
User-Agent: MOR Softswitch
Date: Fri, 30 Jun 2017 20:43:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

And those are the one of the trunk not registering

[2017-06-30 22:48:11] VERBOSE[2782] res_pjsip_logger.c: <--- Transmitting SIP request (590 bytes) to UDP:195.223.107.151:5060 --->
REGISTER sip:sip. macrovoip. it:5060 SIP/2.0
Via: SIP/2.0/UDP 217.19.146.213:5060;rport;branch=z9hG4bKPje6a999fa-1ef4-43db-b207-23f4dc45d98f
From: <sip:390144485210@sip. macrovoip. it>;tag=3866a264-a772-462b-873f-6c61f84629fe
To: <sip:390144485210@sip. macrovoip. it>
Call-ID: fc38b503-79aa-4c14-ad38-24da1d7a6030
CSeq: 34118 REGISTER
Contact: <sip:s@217.19.146.213:5060;line=rlhzgun>
Expires: 60
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Max-Forwards: 70
User-Agent: FPBX-14.0.1rc1.30(13.16.0)
Content-Length: 0


[2017-06-30 22:48:13] DEBUG[2783] config.c: extract uint from [3] in [0, 4294967295] gives [3](0)
[2017-06-30 22:48:13] DEBUG[2783] config.c: extract uint from [0] in [0, 4294967295] gives [0](0)
[2017-06-30 22:48:13] DEBUG[2783] config.c: extract uint from [2] in [0, 4294967295] gives [2](0)
[2017-06-30 22:48:13] DEBUG[2783] res_pjsip.c: 0x2322690: Wrapper created
[2017-06-30 22:48:13] DEBUG[2783] res_pjsip.c: 0x2322690: Set timer to 3000 msec
[2017-06-30 22:48:13] DEBUG[2783] res_pjsip/pjsip_message_ip_updater.c: Re-wrote Contact URI host/port to 192.168.250.5:5060
[2017-06-30 22:48:13] VERBOSE[2783] res_pjsip_logger.c: <--- Transmitting SIP request (472 bytes) to UDP:195.223.107.151:5060 --->
OPTIONS sip:390144485210@sip. macrovoip. it:5060 SIP/2.0
Via: SIP/2.0/UDP 217.19.146.213:5060;rport;branch=z9hG4bKPjff62bccf-ea09-42ae-9587-12169650f8ee
From: <sip:390144485210@192.168.250.5>;tag=58bed95d-964e-45c5-b4cd-c075fc569f4a
To: <sip:390144485210@sip. macrovoip. it>
Contact: <sip:390144485210@217.19.146.213:5060>
Call-ID: 50e8ac62-889b-4d5b-9159-639640c4fcb4
CSeq: 23761 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-14.0.1rc1.30(13.16.0)
Content-Length: 0


[2017-06-30 22:48:13] VERBOSE[2782] res_pjsip_logger.c: <--- Received SIP response (575 bytes) from UDP:195.223.107.151:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.19.146.213:5060;branch=z9hG4bKPjff62bccf-ea09-42ae-9587-12169650f8ee;received=217.19.146.213;rport=26962
From: <sip:390144485210@192.168.250.5>;tag=58bed95d-964e-45c5-b4cd-c075fc569f4a
To: <sip:390144485210@sip. macrovoip. it>;tag=as4eda4e8f
Call-ID: 50e8ac62-889b-4d5b-9159-639640c4fcb4
CSeq: 23761 OPTIONS
Server: MOR Softswitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:195.223.107.151:5060>
Accept: application/sdp
Content-Length: 0

You can notice also that is the same platform (two different providers, but same platform)

The one that work is self-hosted.

I've almost tried every parameter in pjsip trunk configuration

Thank You!

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