Hi All,
Problem:
I have a problem with HT-503 devices (3 off for 3 different locations - 2 of which are remote to me)
I can get incoming routes working, but outgoing routes does not work. i get the generic "all circuits are busy now .....".
I have had this working on a previous system that i was playing with, but decided to start again from scratch to eliminate any potential problems due to "fat fingers" from tinkering too much.
I changed nothing on the HT-503's between setups and was wise enough to keep the old system running, so i can compare the two configs and have gone over and over and over everything i can see that might alter the outcome. I usually like to fix things myself so i learn from it, but this has me stuffed.
Setup:
Incredible PBX 12.0.74 'Incredible PBX Lean GUI'
I have followed the guide "http://wiki.freepbx.org/pages/viewpage.action?pageId=33293313" which is very helpful and pointed out where i went wrong last time.
I now have Stage 1 & 4 working fine from the above guide, that is:
Also, i can get outbound routes working fine with my VOIP account, so i know im doing it right, but when i put the HT-503 trunk into the route it tells me "all circuits busy".
I have tried this from soft-phone, hard-phone and event mobile soft-phone (c-sip-simple on android)
I have tried to stick with CHAN_PJSIP only config (port 5060) when setting up, but i noticed that CHAN_SIP (port 5160) is being called in the logs ... why ? should i convert the PJSIP extensions for the HT-503's to be CHAN_SIP instead ? if yes, then why, this is not what i am seeing in the screenshots on the above guide and in other peoples troubleshooting.
Any Assistance is greatly appreciated, hopefully someone can see something in the debug output below.
Debug output on call:
Problem:
I have a problem with HT-503 devices (3 off for 3 different locations - 2 of which are remote to me)
I can get incoming routes working, but outgoing routes does not work. i get the generic "all circuits are busy now .....".
I have had this working on a previous system that i was playing with, but decided to start again from scratch to eliminate any potential problems due to "fat fingers" from tinkering too much.
I changed nothing on the HT-503's between setups and was wise enough to keep the old system running, so i can compare the two configs and have gone over and over and over everything i can see that might alter the outcome. I usually like to fix things myself so i learn from it, but this has me stuffed.
Setup:
Incredible PBX 12.0.74 'Incredible PBX Lean GUI'
I have followed the guide "http://wiki.freepbx.org/pages/viewpage.action?pageId=33293313" which is very helpful and pointed out where i went wrong last time.
I now have Stage 1 & 4 working fine from the above guide, that is:
- I can dial the extension and get a dial-tone
- Incoming call routing works great.
Also, i can get outbound routes working fine with my VOIP account, so i know im doing it right, but when i put the HT-503 trunk into the route it tells me "all circuits busy".
I have tried this from soft-phone, hard-phone and event mobile soft-phone (c-sip-simple on android)
I have tried to stick with CHAN_PJSIP only config (port 5060) when setting up, but i noticed that CHAN_SIP (port 5160) is being called in the logs ... why ? should i convert the PJSIP extensions for the HT-503's to be CHAN_SIP instead ? if yes, then why, this is not what i am seeing in the screenshots on the above guide and in other peoples troubleshooting.
Any Assistance is greatly appreciated, hopefully someone can see something in the debug output below.
Debug output on call:
Code:
[2016-04-04 15:06:14] SECURITY[2242]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="ChallengeSent",EventTV="2016-04-04T15:06:14.163+1000",Severity="Informational",Service="PJSIP",EventVersion="1",AccountID="106",SessionID="[email protected]",LocalAddress="IPV4/UDP/1.2.3.4/5060",RemoteAddress="IPV4/UDP/1.2.3.106/5060",Challenge=""
[2016-04-04 15:06:14] SECURITY[2242]: res_security_log.c:116 security_event_stasis_cb: SecurityEvent="SuccessfulAuth",EventTV="2016-04-04T15:06:14.173+1000",Severity="Informational",Service="PJSIP",EventVersion="1",AccountID="106",SessionID="[email protected]",LocalAddress="IPV4/UDP/1.2.3.4/5060",RemoteAddress="IPV4/UDP/1.2.3.106/5060",UsingPassword="1"
[2016-04-04 15:06:14] WARNING[2114]: func_cdr.c:352 cdr_write_callback: CDR requires a value (CDR(variable)=value)
Audio is at 19482
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 1.2.3.13:5062:
INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.4:5160;branch=z9hG4bK1d7c7bbf
Max-Forwards: 70
From: <sip:[email protected]:5160>;tag=as58658b2e
To: <sip:[email protected]:5062>
Contact: <sip:[email protected]:5160>
Call-ID: [email protected]:5160
CSeq: 102 INVITE
User-Agent: FPBX-12.0.74(13.7.2)
Date: Mon, 04 Apr 2016 05:06:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 467
v=0
o=root 1527769592 1527769592 IN IP4 1.2.3.4
s=Asterisk PBX 13.7.2
c=IN IP4 1.2.3.4
t=0 0
m=audio 19482 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=ice-ufrag:45b83dce77ab3326541fb9ed5f7769e3
a=ice-pwd:513fca8c3ce1319c48a89cc9010c9354
a=candidate:Ha010164 1 UDP 2130706431 1.2.3.4 19482 typ host
a=candidate:Ha010164 2 UDP 2130706430 1.2.3.4 19483 typ host
a=sendrecv
---
<--- SIP read from UDP:1.2.3.13:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 1.2.3.4:5160;branch=z9hG4bK1d7c7bbf
From: <sip:[email protected]:5160>;tag=as58658b2e
To: <sip:[email protected]:5062>
Call-ID: [email protected]:5160
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V2.0A 1.0.14.1 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:1.2.3.13:5062 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 1.2.3.4:5160;branch=z9hG4bK1d7c7bbf
From: <sip:[email protected]:5160>;tag=as58658b2e
To: <sip:[email protected]:5062>;tag=601623564
Call-ID: [email protected]:5160
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V2.0A 1.0.14.1 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 1.2.3.13:5062:
ACK sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.4:5160;branch=z9hG4bK1d7c7bbf
Max-Forwards: 70
From: <sip:[email protected]:5160>;tag=as58658b2e
To: <sip:[email protected]:5062>;tag=601623564
Contact: <sip:[email protected]:5160>
Call-ID: [email protected]:5160
CSeq: 102 ACK
User-Agent: FPBX-12.0.74(13.7.2)
Content-Length: 0
---
[2016-04-04 15:06:14] WARNING[2209][C-0000008c]: chan_sip.c:23372 handle_response_invite: Received response: "Forbidden" from '<sip:[email protected]:5160>;tag=as58658b2e'
Scheduling destruction of SIP dialog '[email protected]:5160' in 32000 ms (Method: INVITE)