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dor
Registration Date: 01.01.1970
Posts:
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Problem with incoming calls |
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My main business line stopped working with no obvious cause.
Caller just hears short dial tones. On my side phones do ring after some delay, but by that time the caller is already disconnected.
I tried to monitor what's going on using Status screen: while caller is calling (extension is in red color) nothing rings on my side. After 10 seconds call disconnects by itself (extension icon goes green, caller gets short tones), then the phones start ringing. If I answer I obviously hear nothing as the caller is already disconnected.
In the log is see:
Sep 27 16:49:38 VERBOSE[1306] logger.c: -- Got SIP response 481 "Call Leg/Transaction Does Not Exist"
I opened a call in voip.ms support - they used to be highly reliable provider.
Here's what they say:
Zitat: |
A SIp trace indicates that the problem seem to be located at PBXES, or its destination.
When we dial the number, it immediatly reaches our server, and tries to dial the sip uri you have set at pbxes. Pbxes sends back a "Trying" and nothing more, like if it was stuck, call is cancelled about 10 seconds later.
INVITE sip:doronin-0202@pbxes.org SIP/2.0
Via: SIP/2.0/UDP 67.205.74.164:5060;branch=z9hG4bK38b1433e;rport
From: "5146678178" ;tag=as4f7295fa
To:
Contact:
Call-ID: 4a6d81b56f17a45b008caa06267a0316@67.205.74.164
CSeq: 102 INVITE
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "5146678178" ;privacy=off;screen=no
Date: Mon, 28 Sep 2009 18:35:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 411
v=0
o=root 30851 30851 IN IP4 67.205.74.164
s=session
c=IN IP4 67.205.74.164
t=0 0
m=audio 14870 RTP/AVP 0 8 18 3 111 4 101
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:111 G726-32/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called doronin-0202@pbxes.org
ca1*CLI>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.205.74.164:5060;branch=z9hG4bK38b1433e
From: "5146678178" ;tag=as4f7295fa
To:
Call-ID: 4a6d81b56f17a45b008caa06267a0316@67.205.74.164
CSeq: 102 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact:
Content-Length: 0
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P.S. I tried to send incoming call to both registered trunk and SIP URI with the same result.
This post has been edited 1 time(s), it was last edited by dor on 29.09.2009 at 02:48.
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29.09.2009 02:38 |
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Dia
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Registration Date: 03.03.2006
Posts: 1443
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01.10.2009 08:56 |
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dor
Registration Date: 01.01.1970
Posts:
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02.10.2009 04:51 |
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Dia
Premium Account
Registration Date: 03.03.2006
Posts: 1443
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05.10.2009 09:40 |
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mon
Premium Account
Registration Date: 17.10.2007
Posts: 106
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05.10.2009 17:16 |
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cc
Registration Date: 01.01.1970
Posts:
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05.10.2009 21:49 |
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dor
Registration Date: 01.01.1970
Posts:
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RE: Problem with incoming calls |
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Zitat: |
Originally posted by Diafora
• Do you still have it setup as a URI or a registered trunk?
• Can you ask VoIP.ms to send you another trace, now that is working? |
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Here's the new trace. Since then I switched back to registered trunk (I had SIP URI just as an unsuccessful attempt to cure this problem)
INVITE sip:5143330202@64.118.93.76:27436 SIP/2.0
Via: SIP/2.0/UDP 67.205.74.164:5060;branch=z9hG4bK00f21f6d;rport
From: "Steve P " <sip:8777864767@67.205.74.164>;tag=as0721813d
To: <sip:5143330202@64.118.93.76:27436>
Contact: <sip:8777864767@67.205.74.164>
Call-ID: 1cc4258369718d640a4612a1661d287c@67.205.74.164
CSeq: 102 INVITE
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "Steve P " <sip:8777864767@67.205.74.164>;privacy=off;screen=no
Date: Thu, 08 Oct 2009 18:28:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 242
v=0
o=root 30851 30851 IN IP4 67.205.74.164
s=session
c=IN IP4 67.205.74.164
t=0 0
m=audio 13702 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 101108/5143330202
ca1*CLI>
<--- SIP read from 64.118.93.76:27436 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.205.74.164:5060;branch=z9hG4bK00f21f6d;received=67.205.74.164;rport=5060
From: "Steve P " <sip:8777864767@67.205.74.164>;tag=as0721813d
To: <sip:5143330202@64.118.93.76:27436>
Call-ID: 1cc4258369718d640a4612a1661d287c@67.205.74.164
CSeq: 102 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:5143330202@64.118.93.76:27436>
Content-Length: 0
<------------->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 67.205.74.164:5060;branch=z9hG4bK00f21f6d;received=67.205.74.164;rport=5060
From: "Steve P " <sip:8777864767@67.205.74.164>;tag=as0721813d
To: <sip:5143330202@64.118.93.76:27436>;tag=as6a00ffa6
Call-ID: 1cc4258369718d640a4612a1661d287c@67.205.74.164
CSeq: 102 INVITE
User-Agent: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:5143330202@64.118.93.76:27436>
Content-Length: 0
<------------->
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09.10.2009 02:53 |
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Dia
Premium Account
Registration Date: 03.03.2006
Posts: 1443
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09.10.2009 10:10 |
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