TIPS PJSIP Transport, udp 5065

[Gary Vassalotti]

I have obtained some Fanvil X3SP phones, and am trying to get them to work with PIAF...

At one point, I did receive calls from google voice trunk... but could not dial out.. so I switched to pjsip. Now, I can't get the phone to connect. (I did change the ports setting on the phone).

When I run pjsip show endpoints, I get this:

Transport: <TransportId........> <Type> <cos> <tos> <BindAddress....................>
==========================================================================================
Transport: 0.0.0.0-tcp tcp 0 0 0.0.0.0:5160
Transport: 0.0.0.0-udp udp 0 0 0.0.0.0:5160
Transport: incoming-registrations-unused-but-required udp 0 0 0.0.0.0:5065
Transport: transport_tls tls 0 0 0.0.0.0:5061


I don't understand the incoming-registrations line... I don't see port 5065 anywhere in my setups to change or add...

 

[wardmundy]

Here's a good rule of thumb with your PBX. If you don't understand what certain settings do, don't mess with them or you most probably will break something else. In this case, it would be GVSIP support for Google Voice. Stick with chan_sip if you're new to all of this, and we'll get your outbound calling solved. What does the Asterisk CLI show when you attempt to make a call??

 

[Gary Vassalotti]

Here's a good rule of thumb with your PBX. If you don't understand what certain settings do, don't mess with them or you most probably will break something else. In this case, it would be GVSIP support for Google Voice. Stick with chan_sip if you're new to all of this, and we'll get your outbound calling solved. What does the Asterisk CLI show when you attempt to make a call??

Thanks for the quick reply, Ward. I am going to re-flash the raspberry Monday to be sure all settings are back to the original and try again.

 

[wardmundy]

Thanks for the quick reply, Ward. I am going to re-flash the raspberry Monday to be sure all settings are back to the original and try again.

After you do that, follow the upgrade procedure for GVSIP that is documented in this article.

 

[Gary Vassalotti]

Ok, I reflashed the image and did the update procedures in the link you provided. I am pretty sure that google voice works, as I can dial in to the number, and it goes to the voice mail: this shows in the shell window:

> 0x75d45c08 -- Strict RTP learning after remote address set to: 74.125.39.28:19305
[2018-08-21 12:24:58] ERROR[21452]: pjproject:0 <?>: icess0x75d485c ...Error sending STUN request: Network is unreachable
[2018-08-21 12:24:58] ERROR[20820]: pjproject:0 <?>: icess0x75d485c ..Error sending STUN request: Network is unreachable
-- <PJSIP/gvsip1-00000000> Playing 'vm-theperson.ulaw' (language 'en')
-- <PJSIP/gvsip1-00000000> Playing 'digits/7.ulaw' (language 'en')
-- <PJSIP/gvsip1-00000000> Playing 'digits/0.ulaw' (language 'en')
-- <PJSIP/gvsip1-00000000> Playing 'digits/1.ulaw' (language 'en')
-- <PJSIP/gvsip1-00000000> Playing 'vm-isunavail.ulaw' (language 'en')
-- <PJSIP/gvsip1-00000000> Playing 'vm-intro.ulaw' (language 'en')
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'PJSIP/gvsip1-00000000' in macro 'vm'
== Spawn extension (macro-exten-vm, s, 21) exited non-zero on 'PJSIP/gvsip1-00000000' in macro 'exten-vm'
== Spawn extension (ext-local, 701, 2) exited non-zero on 'PJSIP/gvsip1-00000000'
-- Executing [h@ext-local:1] Macro("PJSIP/gvsip1-00000000", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/gvsip1-00000000", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/gvsip1-00000000", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("PJSIP/gvsip1-00000000", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'PJSIP/gvsip1-00000000' in macro 'hangupcall'
== Spawn extension (ext-local, h, 1) exited non-zero on 'PJSIP/gvsip1-00000000'


The phone will not connect / authorize though... When I enter sip show peers, it reports:

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
701 (Unspecified) D Yes Yes A 0 UNKNOWN
1 sip peers [Monitored: 0 online, 1 offline Unmonitored: 0 online, 0 offline]