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Nicolas Olivier

2005-Oct-21 00:58 UTC

[Asterisk-Users] SIP gateway: call hangups afer 3 rings

Hi all,

I've setup an asterisk gw (yyy.yyy.yyy.yyy), connected via IAX2 with another
asterisk (xxx.xxx.xxx.xxx) which acts as a centrex, and via SIP for
incoming/outgoing calls with a provider (zzz.zzz.zzz.zzz).
The problem is that outgoing calls are hanguped after three rings if they are
not answered, and from the debug trace, it seems to be the asterisk gw
who hangups. Apart from that, calls answered before three rings are handled
correctly.
I don't really see what could explain such comportement, and can't find
a "related" sip.conf parameter from the docs, or sample configs.
If anyone has an idea, I've included the related configs and the trace of a
call.

Best regards,
Nicolas Olivier


The gateway is running asterisk 1.0.7.

sip.conf:

[general]
context=default
port=5060
bindaddr=yyy.yyy.yyy.yyy
srvlookup=yes

[provider]
type=friend
host=zzz.zzz.zzz.zzz
port=5060
nat=yes

extensions.conf:

[default]
exten => _x.,1,Dial(SIP/${EXTEN}@provider)
exten => _x.,2,Hangup
exten => _x.,3,Congestion

(...)

Call debug:

     -- Accepting AUTHENTICATED call from xxx.xxx.xxx.xxx requested format = 2,
actual format = 2
     -- Executing Dial("IAX2/centrex-ak@centrex/1",
"SIP/0123456789@provider") in new stack
We're at yyy.yyy.yyy.yyy port 12108
Answering/Requesting with root capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:0123456789@zzz.zzz.zzz.zzz SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5922a0f1;rport
From: "Choco Bobo"
<sip:9876543210@yyy.yyy.yyy.yyy>;tag=as1a492e28
To: <sip:0123456789@zzz.zzz.zzz.zzz>
Contact: <sip:9876543210@yyy.yyy.yyy.yyy>
Call-ID: 6d78077c2452a5ad2a870b584190693c@yyy.yyy.yyy.yyy
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 19 Sep 1980 10:09:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 13764 13764 IN IP4 yyy.yyy.yyy.yyy
s=session
c=IN IP4 yyy.yyy.yyy.yyy
t=0 0
m=audio 12108 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
  (NAT) to zzz.zzz.zzz.zzz:5060
     -- Called 0123456789@b3g


Sip read:
SIP/2.0 100 Trying
Allow: UPDATE,REFER
Call-ID: 6d78077c2452a5ad2a870b584190693c@yyy.yyy.yyy.yyy
Contact: <sip:zzz.zzz.zzz.zzz:5060>
CSeq: 102 INVITE
From: "Choco Bobo"
<sip:9876543210@yyy.yyy.yyy.yyy>;tag=as1a492e28
Server: Cirpack/v4.38f (gw_sip)
To: <sip:0123456789@zzz.zzz.zzz.zzz>
Via: SIP/2.0/UDP
yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1
Content-Length: 0


10 headers, 0 lines


Sip read:
SIP/2.0 183 In band info available
Allow: UPDATE,REFER
Call-ID: 6d78077c2452a5ad2a870b584190693c@yyy.yyy.yyy.yyy
Contact: <sip:zzz.zzz.zzz.zzz:5060>
Content-Type: application/sdp
CSeq: 102 INVITE
From: "Choco Bobo"
<sip:9876543210@yyy.yyy.yyy.yyy>;tag=as1a492e28
Server: Cirpack/v4.38f (gw_sip)
To: <sip:0123456789@zzz.zzz.zzz.zzz>;tag=01-08086-78a18de8-67bc990a2
Via: SIP/2.0/UDP
yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1
Content-Length: 201

v=0
o=cp10 112987934014 112987934014 IN IP4 bbb.bbb.bbb.bbb
s=SIP Call
c=IN IP4 aaa.aaa.aaa.aaa
t=0 0
m=audio 30772 RTP/AVP 0 8
b=AS:64
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=ptime:20

11 headers, 10 lines
Found RTP audio format 0
Found RTP audio format 8
Peer audio RTP is at port aaa.aaa.aaa.aaa:30772
Found description format PCMU
Found description format PCMA
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc
(ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0
(nothing)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0
(nothing)
     -- SIP/b3g-7bfa is making progress passing it to IAX2/centrex-ak@centrex/1
Reliably Transmitting:
CANCEL sip:0123456789@zzz.zzz.zzz.zzz SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5922a0f1;rport
From: "Choco Bobo"
<sip:9876543210@yyy.yyy.yyy.yyy>;tag=as1a492e28
To: <sip:0123456789@zzz.zzz.zzz.zzz>
Contact: <sip:9876543210@yyy.yyy.yyy.yyy>
Call-ID: 6d78077c2452a5ad2a870b584190693c@yyy.yyy.yyy.yyy
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

  (NAT) to zzz.zzz.zzz.zzz:5060
Scheduling destruction of call
'6d78077c2452a5ad2a870b584190693c@yyy.yyy.yyy.yyy' in 15000 ms
   == Spawn extension (default, 0123456789, 4) exited non-zero on
'IAX2/centrex-ak@centrex/1'
     -- Hungup 'IAX2/centrex-ak@centrex/1'


Sip read:
SIP/2.0 200 OK
Call-ID: 6d78077c2452a5ad2a870b584190693c@yyy.yyy.yyy.yyy
CSeq: 102 CANCEL
From: "Choco Bobo"
<sip:9876543210@yyy.yyy.yyy.yyy>;tag=as1a492e28
To: <sip:0123456789@zzz.zzz.zzz.zzz>
Via: SIP/2.0/UDP
yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1
Content-Length: 0

(...)

Nicolas Olivier

2005-Oct-21 02:42 UTC

head link

[Asterisk-Users] SIP gateway: call hangups afer 3 rings

Hi all,

I've setup an asterisk gw (yyy.yyy.yyy.yyy), connected via IAX2 with another
asterisk (xxx.xxx.xxx.xxx) which acts as a centrex, and via SIP for
incoming/outgoing calls with a provider (zzz.zzz.zzz.zzz).
The problem is that outgoing calls are hanguped after three rings if they are
not answered, and from the debug trace, it seems to be the asterisk gw
who hangups. Apart from that, calls answered before three rings are handled
correctly.
I don't really see what could explain such comportement, and can't find
a "related" sip.conf parameter from the docs, or sample configs.
If anyone has an idea, I've included the related configs and the trace of a
call.

Best regards,
Nicolas Olivier


The gateway is running asterisk 1.0.7.

sip.conf:

[general]
context=default
port=5060
bindaddr=yyy.yyy.yyy.yyy
srvlookup=yes

[provider]
type=friend
host=zzz.zzz.zzz.zzz
port=5060
nat=yes

extensions.conf:

[default]
exten => _x.,1,Dial(SIP/${EXTEN}@provider)
exten => _x.,2,Hangup
exten => _x.,3,Congestion

(...)

Call debug:

     -- Accepting AUTHENTICATED call from xxx.xxx.xxx.xxx requested format = 2,
actual format = 2
     -- Executing Dial("IAX2/centrex-ak@centrex/1",
"SIP/0123456789@provider") in new stack
We're at yyy.yyy.yyy.yyy port 12108
Answering/Requesting with root capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:0123456789@zzz.zzz.zzz.zzz SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5922a0f1;rport
From: "Choco Bobo"
<sip:9876543210@yyy.yyy.yyy.yyy>;tag=as1a492e28
To: <sip:0123456789@zzz.zzz.zzz.zzz>
Contact: <sip:9876543210@yyy.yyy.yyy.yyy>
Call-ID: 6d78077c2452a5ad2a870b584190693c@yyy.yyy.yyy.yyy
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 19 Sep 1980 10:09:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 13764 13764 IN IP4 yyy.yyy.yyy.yyy
s=session
c=IN IP4 yyy.yyy.yyy.yyy
t=0 0
m=audio 12108 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
  (NAT) to zzz.zzz.zzz.zzz:5060
     -- Called 0123456789@b3g


Sip read:
SIP/2.0 100 Trying
Allow: UPDATE,REFER
Call-ID: 6d78077c2452a5ad2a870b584190693c@yyy.yyy.yyy.yyy
Contact: <sip:zzz.zzz.zzz.zzz:5060>
CSeq: 102 INVITE
From: "Choco Bobo"
<sip:9876543210@yyy.yyy.yyy.yyy>;tag=as1a492e28
Server: Cirpack/v4.38f (gw_sip)
To: <sip:0123456789@zzz.zzz.zzz.zzz>
Via: SIP/2.0/UDP
yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1
Content-Length: 0


10 headers, 0 lines


Sip read:
SIP/2.0 183 In band info available
Allow: UPDATE,REFER
Call-ID: 6d78077c2452a5ad2a870b584190693c@yyy.yyy.yyy.yyy
Contact: <sip:zzz.zzz.zzz.zzz:5060>
Content-Type: application/sdp
CSeq: 102 INVITE
From: "Choco Bobo"
<sip:9876543210@yyy.yyy.yyy.yyy>;tag=as1a492e28
Server: Cirpack/v4.38f (gw_sip)
To: <sip:0123456789@zzz.zzz.zzz.zzz>;tag=01-08086-78a18de8-67bc990a2
Via: SIP/2.0/UDP
yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1
Content-Length: 201

v=0
o=cp10 112987934014 112987934014 IN IP4 bbb.bbb.bbb.bbb
s=SIP Call
c=IN IP4 aaa.aaa.aaa.aaa
t=0 0
m=audio 30772 RTP/AVP 0 8
b=AS:64
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=ptime:20

11 headers, 10 lines
Found RTP audio format 0
Found RTP audio format 8
Peer audio RTP is at port aaa.aaa.aaa.aaa:30772
Found description format PCMU
Found description format PCMA
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc
(ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0
(nothing)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0
(nothing)
     -- SIP/b3g-7bfa is making progress passing it to IAX2/centrex-ak@centrex/1
Reliably Transmitting:
CANCEL sip:0123456789@zzz.zzz.zzz.zzz SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5922a0f1;rport
From: "Choco Bobo"
<sip:9876543210@yyy.yyy.yyy.yyy>;tag=as1a492e28
To: <sip:0123456789@zzz.zzz.zzz.zzz>
Contact: <sip:9876543210@yyy.yyy.yyy.yyy>
Call-ID: 6d78077c2452a5ad2a870b584190693c@yyy.yyy.yyy.yyy
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

  (NAT) to zzz.zzz.zzz.zzz:5060
Scheduling destruction of call
'6d78077c2452a5ad2a870b584190693c@yyy.yyy.yyy.yyy' in 15000 ms
   == Spawn extension (default, 0123456789, 4) exited non-zero on
'IAX2/centrex-ak@centrex/1'
     -- Hungup 'IAX2/centrex-ak@centrex/1'


Sip read:
SIP/2.0 200 OK
Call-ID: 6d78077c2452a5ad2a870b584190693c@yyy.yyy.yyy.yyy
CSeq: 102 CANCEL
From: "Choco Bobo"
<sip:9876543210@yyy.yyy.yyy.yyy>;tag=as1a492e28
To: <sip:0123456789@zzz.zzz.zzz.zzz>
Via: SIP/2.0/UDP
yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1
Content-Length: 0

(...)

asterisk users - Oct 2005 - SIP gateway: call hangups afer 3 rings

[Asterisk-Users] SIP gateway: call hangups afer 3 rings

[Asterisk-Users] SIP gateway: call hangups afer 3 rings