PBXes » English » Providers » RE: Problem with incoming calls
Print Page | Recommend to Friend | Add Thread to Favorites
Post New Thread Post Reply
Author
Post « Previous Thread | Next Thread »
dor


Registration Date: 01.01.1970
Posts:

Problem with incoming calls Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

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


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 01:48.
29.09.2009 01:38 doronin is offline Search for Posts by doronin Add doronin to your Buddy List
Dia
Premium Account


Registration Date: 03.03.2006
Posts: 1443
RE: Problem with incoming calls Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

Well, the trace certainly indicates a couple of issues:

The SIP URI their SIP Proxy is trying to send the inbound call to, is doronin-0202@pbxes.org. Is such an Inbound Route setup in your PBXes account?

The To: as well as the Contact: fields on the Invite are blank, and as far as I know, "the Contact field must be present and contain exactly one SIP or SIPS URI in any request that can result in the establishment of a dialog", according to RFC3261.

Will this voip.ms trunk accept calls, when you set it up on a simple SIP UA? (e.g. ATA, IP-phone, soft-phone)
01.10.2009 07:56 Diafora is offline Search for Posts by Diafora Add Diafora to your Buddy List
dor


Registration Date: 01.01.1970
Posts:

RE: Problem with incoming calls Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

Inbound route is there, and it shouldn't be the cause as originally the setup used registered trunk, not SIP URI. It worked OK, few days ago it broke, and I switched to SIP URI in attempt to troubleshoot the problem.

UPDATE: somehow it's back in track since yesterday. voip.ms said they didn't do anything, so I guess it'll stay a mistery...

This post has been edited 1 time(s), it was last edited by dor on 04.10.2009 at 18:55.
02.10.2009 03:51 doronin is offline Search for Posts by doronin Add doronin to your Buddy List
Dia
Premium Account


Registration Date: 03.03.2006
Posts: 1443
RE: Problem with incoming calls Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

• 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?
05.10.2009 08:40 Diafora is offline Search for Posts by Diafora Add Diafora to your Buddy List
mon
Premium Account


Registration Date: 17.10.2007
Posts: 109
RE: Problem with incoming calls Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

Again same problem what happened few day before, all devices failed to register.

Request Pbxes to look into this frequent failure.
05.10.2009 16:16 montoo is offline Search for Posts by montoo Add montoo to your Buddy List
cc


Registration Date: 01.01.1970
Posts:

RE: Problem with incoming calls Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

I am also having issues with incoming calls today - I just get dead air... I have temporarily forwarded the DID to my cell phone.

pbxes, please figure out what is going on - your reliability has not been very good lately.

I missed 11 calls, thankfully most were from the same person trying to get through. The problems started somewhere between 11:28am and 12:20pm PST.

update: seems to be working fine now...

This post has been edited 2 time(s), it was last edited by cc on 05.10.2009 at 23:36.
05.10.2009 20:49 cc is offline Search for Posts by cc Add cc to your Buddy List
dor


Registration Date: 01.01.1970
Posts:

RE: Problem with incoming calls Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

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?



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


<------------->
09.10.2009 01:53 doronin is offline Search for Posts by doronin Add doronin to your Buddy List
Dia
Premium Account


Registration Date: 03.03.2006
Posts: 1443
RE: Problem with incoming calls Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

As it is clearly shown on the traces, the "To:" and the "Contact:" fields, are properly populated in the "Invite", "100 Trying" as well as the "180 Ringing" messages.

This was most probably the reason incoming calls to your VoIP.ms DID were not being routed to PBXes, and for your callers to listen to short tones.

The VoIP.ms issue is totally unrelated to the general DNS issue, PBXes faced yesterday, which now seems to have been resolved.
09.10.2009 09:10 Diafora is offline Search for Posts by Diafora Add Diafora to your Buddy List
 
Post New Thread Post Reply
Go to:

Powered by Burning Board Lite 1.0.2 © 2001-2004 WoltLab GmbH
English Translation by Satelk