Hello,
I have installed asterisk 1.6 (on centos) and somehow there is a delay in answer. I show the logs, i think it will make clear everything however I do not have timestamps for the logs.
The system is NATed
That happens when someone calls the IVR (from asterisk console):
== Using SIP RTP CoS mark 5
– Executing [xxxxxxxxxx@DID_trunk_1:1] Goto(“SIP/trunk_1-00000009”, “default,7900,1”) in new stack
– Goto (default,7900,1)
– Executing [7900@default:1] Goto(“SIP/trunk_1-00000009”, “voicemenu-custom-2,s,1”) in new stack
– Goto (voicemenu-custom-2,s,1)
– Executing [s@voicemenu-custom-2:1] NoOp(“SIP/trunk_1-00000009”, “IVR_hu”) in new stack
– Executing [s@voicemenu-custom-2:2] Answer(“SIP/trunk_1-00000009”, “”) in new stack <—
– Executing [s@voicemenu-custom-2:3] Set(“SIP/trunk_1-00000009”, “CHANNEL(language)=hu”) in new stack
– Executing [s@voicemenu-custom-2:4] Set(“SIP/trunk_1-00000009”, “modify=0”) in new stack
– Executing [s@voicemenu-custom-2:5] AGI(“SIP/trunk_1-00000009”, “parking/checkuser2.php”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/parking/checkuser2.php
parking/checkuser2.php: Agi started
parking/checkuser2.php: xxxxxxxxxxxx
parking/checkuser2.php: <SIP/trunk_1-00000009>-http://192.168.0.1:81/IVR/mParking_IVR_checkUser.php?callerID=xxxxxxxxxxx@
parking/checkuser2.php: <SIP/trunk_1-00000009>-NextOGM=UserNotRegistered
– <SIP/trunk_1-00000009>AGI Script parking/checkuser2.php completed, returning 0
– Executing [s@voicemenu-custom-2:6] GotoIf(“SIP/trunk_1-00000009”, “0?900”) in new stack
– Executing [s@voicemenu-custom-2:7] GotoIf(“SIP/trunk_1-00000009”, “0?100”) in new stack
– Executing [s@voicemenu-custom-2:8] GotoIf(“SIP/trunk_1-00000009”, “0?200”) in new stack
– Executing [s@voicemenu-custom-2:9] Playback(“SIP/trunk_1-00000009”, “UserNotRegistered”) in new stack
– <SIP/trunk_1-00000009> Playing ‘UserNotRegistered.slin’ (language ‘hu’)
– Executing [s@voicemenu-custom-2:10] GotoIf(“SIP/trunk_1-00000009”, “0?1000”) in new stack
– Executing [s@voicemenu-custom-2:11] GotoIf(“SIP/trunk_1-00000009”, “1?1000”) in new stack
– Goto (voicemenu-custom-2,s,1000)
– Executing [s@voicemenu-custom-2:1000] Playback(“SIP/trunk_1-00000009”, “GoodBye”) in new stack
– <SIP/trunk_1-00000009> Playing ‘GoodBye.slin’ (language ‘hu’)
– Executing [s@voicemenu-custom-2:1001] Hangup(“SIP/trunk_1-00000009”, “”) in new stack
== Spawn extension (voicemenu-custom-2, s, 1001) exited non-zero on ‘SIP/trunk_1-00000009’
Sadly it has no timestamps, but at the asnwer command (<–) it waits for something, but I can’t guess what.
Here is the SIP log between asnwer egy set channel:
– Executing [s@voicemenu-custom-2:2] Answer(“SIP/trunk_1-0000000b”, “”) in new stack
Audio is at 195.70.62.197 port 13770
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x1 (g723) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (NAT) to 195.56.192.1:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.56.192.1:5060;branch=z9hG4bK-6B63-381A09;received=195.56.192.1
From: sip:36205222947@datanet.hu;user=phone;tag=15813-JX-0094dfff-118321eb6
To: sip:3619999798@195.70.62.197;user=phone;tag=as3205f41b
Call-ID: 15813-PE-0094dffe-22895b751@datanet.hu
CSeq: 8847380 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:3619999798@195.70.62.197
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 875985104 875985104 IN IP4 195.70.62.197
s=Asterisk PBX 1.6.2.12
c=IN IP4 195.70.62.197
t=0 0
m=audio 13770 RTP/AVP 8 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
<— SIP read from UDP:195.56.192.1:5060 —>
ACK sip:3619999798@195.70.62.197 SIP/2.0
Call-ID: 15813-PE-0094dffe-22895b751@datanet.hu
Contact: sip:195.56.192.1:5060
CSeq: 8847380 ACK
From: sip:36205222947@datanet.hu;user=phone;tag=15813-JX-0094dfff-118321eb6
Max-Forwards: 30
To: sip:3619999798@195.70.62.197;user=phone;tag=as3205f41b
User-Agent: Cirpack/v4.42q (gw_sip)
Via: SIP/2.0/UDP 195.56.192.1:5060;branch=z9hG4bK-5085-381A0A
Content-Length: 0
<------------->
— (10 headers 0 lines) —
– Executing [s@voicemenu-custom-2:3] Set(“SIP/trunk_1-0000000b”, “CHANNEL(language)=hu”) in new stack
I though core set debug 9 would show something, but nothing more.
Any guess?
Thank you in advance!
Thomas