Hello,
Currently I am using Freepbx 15.0.24 and for my own reasons I changed PJSIP ports to 5080 and I have UDP & TCP enabled. I have many chan_sip clients cooperating flawlessly with Inbound and Outbound routes, ring groups, behind NAT etc including another IAX2 server and some SIP trunks in both of sides. I’m just giving a taste of my office installation and what I am trying to say is that all this works almost perfectly.
My problem is regarding PJSIP extensions. Let’s say you create one (or more) pjsip extensions and you hook on them two new phones. Devices seems registered correctly and you are able to make calls through them to other chan_sip extensions as well trunks etc BUT you cannot call pjsip extensions to each other AND no chan_sip extension can reach them. In other words, no incoming calls. And always gives you a “SIP/2.0 503 Service Unavailable”
Any idea???
<--- Transmitting (NAT) to 192.168.0.17:1024 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.17:1024;branch=z9hG4bK-9dda4831;received=192.168.0.17;rport=1024
From: "Anonymous" <sip:[email protected]>;tag=8be1b21c59542310o0
To: "Office Wireless #251" <sip:[email protected]>;tag=as71f60d9d
Call-ID: 63b380c-919ed5e0@localhost
CSeq: 102 INVITE
Server: FPBX-15.0.24(16.21.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
P-Asserted-Identity: "DESK 01 #251" <sip:[email protected]>
Content-Length: 0
<------------>
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dial-one:56] ExecIf("SIP/201-00000026", "0?MacroExit()") in new stack
-- Executing [s@macro-dial-one:57] ExecIf("SIP/201-00000026", "0?Set(DIALSTATUS=)") in new stack
-- Executing [s@macro-dial-one:58] GosubIf("SIP/201-00000026", "0?s-CHANUNAVAIL,1()") in new stack
-- Executing [s@macro-dial-one:59] MacroExit("SIP/201-00000026", "") in new stack
-- Executing [s@macro-exten-vm:15] Set("SIP/201-00000026", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:16] GosubIf("SIP/201-00000026", "0?docfu,1()") in new stack
-- Executing [s@macro-exten-vm:17] GosubIf("SIP/201-00000026", "0?docfb,1()") in new stack
-- Executing [s@macro-exten-vm:18] Set("SIP/201-00000026", "DIALSTATUS=CHANUNAVAIL") in new stack
-- Executing [s@macro-exten-vm:19] ExecIf("SIP/201-00000026", "0?MacroExit()") in new stack
-- Executing [s@macro-exten-vm:20] GotoIf("SIP/201-00000026", "1?s-CHANUNAVAIL,1") in new stack
-- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-exten-vm:1] GotoIf("SIP/201-00000026", "0?exit,1") in new stack
-- Executing [s-CHANUNAVAIL@macro-exten-vm:2] PlayTones("SIP/201-00000026", "congestion") in new stack
-- Executing [s-CHANUNAVAIL@macro-exten-vm:3] Congestion("SIP/201-00000026", "10") in new stack
<--- Reliably Transmitting (NAT) to 192.168.0.17:1024 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.17:1024;branch=z9hG4bK-9dda4831;received=192.168.0.17;rport=1024
From: "Anonymous" <sip:[email protected]>;tag=8be1b21c59542310o0
To: "Office Wireless #251" <sip:[email protected]>;tag=as71f60d9d
Call-ID: 63b380c-919ed5e0@localhost
CSeq: 102 INVITE
Server: FPBX-15.0.24(16.21.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
Content-Length: 0
<------------>
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on 'SIP/201-00000026' in macro 'exten-vm'
== Spawn extension (from-internal, 251, 3) exited non-zero on 'SIP/201-00000026'
-- Executing [h@from-internal:1] Macro("SIP/201-00000026", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/201-00000026", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/201-00000026", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("SIP/201-00000026", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/201-00000026' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/201-00000026'
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/201-00000026
<--- SIP read from UDP:192.168.0.17:1024 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.17:1024;branch=z9hG4bK-9dda4831
From: "Anonymous" <sip:[email protected]>;tag=8be1b21c59542310o0
To: "Office Wireless #251" <sip:[email protected]>;tag=as71f60d9d
Call-ID: 63b380c-919ed5e0@localhost
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="201",realm="asterisk",nonce="75bf1ced",uri="sip:[email protected]",algorithm=MD5,response="26d12a7906c23cbbc87b4299df16a839"
Contact: "Anonymous" <sip:[email protected]:1024>
User-Agent: Cisco/SPA525G2-7.6.2e
Content-Length: 0