[asterisk-users] channel not hungup (zombie?) so call limit not reset to zero
Rizwan Hisham
rizwanhasham at gmail.com
Fri Aug 24 06:25:06 CDT 2007
here is the sip debug for the channel. Before reading the sip debug i want
ot tell you that user is using "Telco Systems AC-211 v4.50.27" adapter. sip
sdebug shows that asterisk is trying to send the initial invite but there is
no response from the user (after registration, user dies, no single
response). So maybe there is some kind of network issue (NAT) or there is
something wrong with the Telco Systems adapter. The stuck channels are still
there:
IP crunch 260bca1e59e 00102/00000 unkn No Init: INVITE
IP crunch 350d1a6e2b1 00102/00000 unkn No Init: INVITE
and "core show channels" show 0 active calls
*CLI> core show channels
Channel Location State Application(Data)
0 active channels
0 active calls
SIP DEBUG
Audio is at 64.182.161.2 port 10678
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 72.73.66.175:50069:
INVITE sip:crunch at 192.168.1.41:9060 SIP/2.0
Via: SIP/2.0/UDP 64.182.161.2:9060;branch=z9hG4bK473eaef8;rport
From: "adf xyz" <sip:12129339038 at 64.182.161.2:9060>;tag=as22ac8da7
To: <sip:crunch at 192.168.1.41:9060>
Contact: <sip:12129339038 at 64.182.161.2:9060>
Call-ID: 2a02e3804ba80e6f603bc427274193d7 at 64.182.161.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 24 Aug 2007 09:54:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 334
v=0
o=root 28236 28236 IN IP4 64.182.161.2
s=session
c=IN IP4 64.182.161.2
t=0 0
m=audio 10678 RTP/AVP 0 8 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #1 (NAT) to 72.73.66.175:50069:
INVITE sip:crunch at 192.168.1.41:9060 SIP/2.0
Via: SIP/2.0/UDP 64.182.161.2:9060;branch=z9hG4bK473eaef8;rport
From: "adf xyz" <sip:12129339038 at 64.182.161.2:9060>;tag=as22ac8da7
To: <sip:crunch at 192.168.1.41:9060>
Contact: <sip:12129339038 at 64.182.161.2:9060>
Call-ID: 2a02e3804ba80e6f603bc427274193d7 at 64.182.161.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 24 Aug 2007 09:54:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 334
v=0
o=root 28236 28236 IN IP4 64.182.161.2
s=session
c=IN IP4 64.182.161.2
t=0 0
m=audio 10678 RTP/AVP 0 8 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #2 (NAT) to 72.73.66.175:50069:
INVITE sip:crunch at 192.168.1.41:9060 SIP/2.0
Via: SIP/2.0/UDP 64.182.161.2:9060;branch=z9hG4bK473eaef8;rport
From: "adf xyz" <sip:12129339038 at 64.182.161.2:9060>;tag=as22ac8da7
To: <sip:crunch at 192.168.1.41:9060>
Contact: <sip:12129339038 at 64.182.161.2:9060>
Call-ID: 2a02e3804ba80e6f603bc427274193d7 at 64.182.161.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 24 Aug 2007 09:54:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 334
v=0
o=root 28236 28236 IN IP4 64.182.161.2
s=session
c=IN IP4 64.182.161.2
t=0 0
m=audio 10678 RTP/AVP 0 8 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #3 (NAT) to 72.73.66.175:50069:
INVITE sip:crunch at 192.168.1.41:9060 SIP/2.0
Via: SIP/2.0/UDP 64.182.161.2:9060;branch=z9hG4bK473eaef8;rport
From: "adf xyz" <sip:12129339038 at 64.182.161.2:9060>;tag=as22ac8da7
To: <sip:crunch at 192.168.1.41:9060>
Contact: <sip:12129339038 at 64.182.161.2:9060>
Call-ID: 2a02e3804ba80e6f603bc427274193d7 at 64.182.161.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 24 Aug 2007 09:54:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 334
v=0
o=root 28236 28236 IN IP4 64.182.161.2
s=session
c=IN IP4 64.182.161.2
t=0 0
m=audio 10678 RTP/AVP 0 8 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #4 (NAT) to 72.73.66.175:50069:
INVITE sip:crunch at 192.168.1.41:9060 SIP/2.0
Via: SIP/2.0/UDP 64.182.161.2:9060;branch=z9hG4bK473eaef8;rport
From: "adf xyz" <sip:12129339038 at 64.182.161.2:9060>;tag=as22ac8da7
To: <sip:crunch at 192.168.1.41:9060>
Contact: <sip:12129339038 at 64.182.161.2:9060>
Call-ID: 2a02e3804ba80e6f603bc427274193d7 at 64.182.161.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 24 Aug 2007 09:54:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 334
v=0
o=root 28236 28236 IN IP4 64.182.161.2
s=session
c=IN IP4 64.182.161.2
t=0 0
m=audio 10678 RTP/AVP 0 8 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #5 (NAT) to 72.73.66.175:50069:
INVITE sip:crunch at 192.168.1.41:9060 SIP/2.0
Via: SIP/2.0/UDP 64.182.161.2:9060;branch=z9hG4bK473eaef8;rport
From: "adf xyz" <sip:12129339038 at 64.182.161.2:9060>;tag=as22ac8da7
To: <sip:crunch at 192.168.1.41:9060>
Contact: <sip:12129339038 at 64.182.161.2:9060>
Call-ID: 2a02e3804ba80e6f603bc427274193d7 at 64.182.161.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 24 Aug 2007 09:54:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 334
v=0
o=root 28236 28236 IN IP4 64.182.161.2
s=session
c=IN IP4 64.182.161.2
t=0 0
m=audio 10678 RTP/AVP 0 8 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
On 8/23/07, Rizwan Hisham <rizwanhasham at gmail.com> wrote:
>
> im having a strange problem related to call-limit for peers. well im not
> sure if its related to call-limmit or not. Bottom line is:
>
> I call a user A, from user B. user B hears silence, untill it goes to
> voicemail. when user B hangsup. user B's call limit is reset to 0 but user
> A's call limit is not reset.strange thing is user A's status on cli is
> shown as NOANSWER, while user B did not even hear ringing. when i use "sip
> show channels" command, it shows me a channel for user A like below:
>
> crunch 30d926c1055 00102/00000 unkn No Init: INVITE
>
> It stays in INVITE state unless i restart my asterisk server. when i
> restart the channel is clear (ofcorse)
>
> So my guess is, its a zombiee channel which asterisk forgot to hangup.
> WHY? i dont know, maybe there is a problem in sip signalling due to which
> asterisk didnt recieve the bye signal in the first place or maybe its
> asterisk fault totally.
>
> So because it is not hungup by asterisk thats why its call limit is not
> reset to zero. I dont have sip debug for this problem yet, i'll post it
> later when i have it. meanwhile if somebody has experienced a similar
> problem and has successfully fixed it, then plz share my burden and help me.
>
> --
> Best Regards
> Rizwan Hisham
> Software Engineer
> Axvoice Inc.
> www.axvoice.com
--
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
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