Forwarding Issue

Asterisk Asterisk SIP
1

hello I change my mobile phone settings to forward calls from 076160xxxx to 011201xxxx. 011201xxxx is my Sip trunk number. It worked successfully and the IVR was playing. My IVR has the option, when the caller presses 8, it will be forwarded to 078773xxxx. At this point, I called 076160xxxx from my other number and I heard the IVR background voice. Then I pressed 8. The call hung up and the following log was shown in the CLI.

 Executing [8@annoucement_2:3] Dial("PJSIP/my_sip_trunk-00000009", "PJSIP/0787733966@my_sip_trunk") in new stack
    -- Called PJSIP/0787733966@my_sip_trunk
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'PJSIP/my_sip_trunk-00000009' status is 'CHANUNAVAIL'

What is the issue? And How fix this?

2

You need to do basic troubleshooting to narrow it down:

  1. Does Asterisk show my_sip_trunk as available in “pjsip show endpoints”?
  2. What does the SIP traffic from “pjsip set logger on” show?
3
  1. Yes. it is available in “pjsip show endpoints”?
 Endpoint:  my_sip_trunk                                         Not in use    0 of inf
        Aor:  my_sip_trunk                                       0
      Contact:  my_sip_trunk/sip:172.20.10.86              439434fad1 NonQual         nan
   Identify:  my_sip_trunk/my_sip_trunk
        Match: 172.20.10.86/32

And this is logger details

 Executing [8@annoucement_2:3] Dial("PJSIP/my_sip_trunk-00000000", "PJSIP/0787733966@my_sip_trunk") in new stack
    -- Called PJSIP/0787733966@my_sip_trunk
<--- Transmitting SIP request (1033 bytes) to UDP:172.20.10.86:5060 --->
INVITE sip:0787733966@172.20.10.86 SIP/2.0
Via: SIP/2.0/UDP 172.20.10.100:5060;rport;branch=z9hG4bKPj99c86c64-24c6-421d-9473-e86b7dfa6766
From: <sip:112019510@172.20.10.100>;tag=6fdc9ef6-5884-4531-aa21-8fce378f2f4b
To: <sip:0787733966@172.20.10.86>
Contact: <sip:112019510@172.20.10.100:5060>
Call-ID: a25c7f4c-e746-4e16-9200-7fb5351f5303
CSeq: 28572 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Diversion: <sip:717306998@172.20.10.100>;reason=unconditional
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Type: application/sdp
Content-Length:   294

v=0
o=- 351943068 351943068 IN IP4 172.20.10.100
s=Asterisk
c=IN IP4 172.20.10.100
t=0 0
m=audio 11612 RTP/AVP 0 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

<--- Received SIP response (556 bytes) from UDP:172.20.10.86:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.20.10.100:5060;rport;branch=z9hG4bKPj99c86c64-24c6-421d-9473-e86b7dfa6766
From: <sip:112019510@172.20.10.100>;tag=6fdc9ef6-5884-4531-aa21-8fce378f2f4b
To: <sip:0787733966@172.20.10.86>;tag=1c888132780
Call-ID: a25c7f4c-e746-4e16-9200-7fb5351f5303
CSeq: 28572 INVITE
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000 - MSBG/v.6.40A.029.008
Content-Length: 0


<--- Received SIP response (612 bytes) from UDP:172.20.10.86:5060 --->
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 172.20.10.100:5060;rport;branch=z9hG4bKPj99c86c64-24c6-421d-9473-e86b7dfa6766
From: <sip:112019510@172.20.10.100>;tag=6fdc9ef6-5884-4531-aa21-8fce378f2f4b
To: <sip:0787733966@172.20.10.86>;tag=1c888132780
Call-ID: a25c7f4c-e746-4e16-9200-7fb5351f5303
CSeq: 28572 INVITE
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000 - MSBG/v.6.40A.029.008
Reason: Q.850 ;cause=21 ;text="local"
Content-Length: 0


<--- Transmitting SIP request (403 bytes) to UDP:172.20.10.86:5060 --->
ACK sip:0787733966@172.20.10.86 SIP/2.0
Via: SIP/2.0/UDP 172.20.10.100:5060;rport;branch=z9hG4bKPj99c86c64-24c6-421d-9473-e86b7dfa6766
From: <sip:112019510@172.20.10.100>;tag=6fdc9ef6-5884-4531-aa21-8fce378f2f4b
To: <sip:0787733966@172.20.10.86>;tag=1c888132780
Call-ID: a25c7f4c-e746-4e16-9200-7fb5351f5303
CSeq: 28572 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'PJSIP/my_sip_trunk-00000000' status is 'CHANUNAVAIL'
<--- Received SIP request (612 bytes) from UDP:172.20.10.86:5060 --->
BYE sip:172.20.10.100:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.10.86;branch=z9hG4bKac1520537467
Max-Forwards: 70
From: "761604153" <sip:0761604153@172.20.10.86>;tag=1c2058900502
To: <sip:0112019510@172.20.10.100;user=phone>;tag=02bf8051-f1ce-4816-8e6a-884b278f2317
Call-ID: 2058846314662024215940@172.20.10.86
CSeq: 2 BYE
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-Mediant 1000 - MSBG/v.6.40A.029.008
Reason: Q.850 ;cause=16 ;text="local"
Content-Length: 0


<--- Transmitting SIP response (371 bytes) to UDP:172.20.10.86:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.20.10.86;rport=5060;received=172.20.10.86;branch=z9hG4bKac1520537467
Call-ID: 2058846314662024215940@172.20.10.86
From: "761604153" <sip:0761604153@172.20.10.86>;tag=1c2058900502
To: <sip:0112019510@172.20.10.100;user=phone>;tag=02bf8051-f1ce-4816-8e6a-884b278f2317
CSeq: 2 BYE
Server: Asterisk PBX 20.5.0
Content-Length:  0


  == MixMonitor close filestream (mixed)
  == End MixMonitor Recording PJSIP/my_sip_trunk-00000000
4

The AudioCodes gateway responded with:

and a reason of:

So, you need to figure out what it doesn’t like.

5

After I change my ‘from_user’ number, it was success.

1 Like
6

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.