10-16-2009 03:38 AM - edited 03-17-2019 09:51 PM
Hello to all! We have a bit of a problem
Our customer has a geographically spread voip. Central site and spokes. Some spokes have local phone ports only and are connected via h.323 peers, some have local pbx connected via sip and gateways there only do number manipulation in sip-to-sip calls.
The lineup is like this: Panasonic TDA200 through E1 Pri (net5 type) to Cisco3825 to Cisco2811 through SIP to Panasonic PBX with SIP card. The problem is that when remote user on sip-emabled panasonic is busy, local user calling him on tda200 trough E1 to 3825 and then through vpn tunnel to 2811 waits for ~30 secs and then gets reorder instead of getting immediate busy tone. When the user is not busy the communication establishes but I have no info about delays.
I have enabled gateway accounting and debugged isdn q931. Dialing panasonic sends 3 digits enblock and then adds 1 in overlap, it cannot do otherwise. That's ok. What I cannot understand is that gw accounting reasonably shows that the call started and ended at the same time and code was user busy. But the gw sends to originating ISDN channel DISCONNECT message only some 30 secs later with normal call clearing code.
To our knowledge this only happens with sip-connected voip spokes, in h.323 there is no such thing.
What could cause that?
debug info:
****************************************
*Oct 16 10:03:50.779: ISDN Se0/3/0:15 Q931: RX <- SETUP pd = 8 callref = 0x0022
Bearer Capability i = 0x9090A3
Standard = CCITT
Transfer Capability = 3.1kHz Audio
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA1839E
Preferred, Channel 30
Progress Ind i = 0x8183 - Origination address is non-ISDN
Calling Party Number i = 0x2180, '328'
Plan:ISDN, Type:National
Called Party Number i = 0x81, '600'
Plan:ISDN, Type:Unknown
*Oct 16 10:03:50.783: ISDN Se0/3/0:15 Q931: TX -> SETUP_ACK pd = 8 callref = 0x8022
Channel ID i = 0xA9839E
Exclusive, Channel 30
*Oct 16 10:03:50.951: ISDN Se0/3/0:15 Q931: RX <- INFORMATION pd = 8 callref = 0x0022
Called Party Number i = 0x81, '8'
Plan:ISDN, Type:Unknown
*Oct 16 10:03:50.959: ISDN Se0/3/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8022
Kievskaya#
*Oct 16 10:03:51.287: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 2, ConnectionId 9B275DCB97211DE95F20024C4168480, SetupTime *10:03:50.957 UTC Fri Oct 16 2009, PeerAddress 6008, PeerSubAddress , DisconnectCause 11 , DisconnectText user busy (17), ConnectTime *10:03:51.287 UTC Fri Oct 16 2009, DisconnectTime *10:03:51.287 UTC Fri Oct 16 2009, CallOrigin 1, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 0, ReceiveBytes 0
*Oct 16 10:03:51.287: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:10/16/2009 10:03:50.955,cgn:328,cdn:6008,frs:0,fid:57757,fcid:9B275DCB97211DE95F20024C4168480,legID:E1B8,bguid:09B275DCB97211DE95F20024C4168480
Kievskaya#
*Oct 16 10:04:21.291: ISDN Se0/3/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x8022
Cause i = 0x8290 - Normal call clearing
*Oct 16 10:04:21.323: ISDN Se0/3/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x0022
*Oct 16 10:04:21.327: ISDN Se0/3/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x8022
peers configuration:
3825:
dial-peer voice 14 pots
destination-pattern [123]..
port 0/3/0:15
forward-digits all
dial-peer voice 117 voip
description outg to Kursk SIP
destination-pattern 60..
session protocol sipv2
session target ipv4:172.17.100.9
dtmf-relay h245-alphanumeric
codec g729br8
2811:
dial-peer voice 11 voip
description outgoing to local SIP PBX
translation-profile outgoing IP_to_Local
destination-pattern 60..
session protocol sipv2
session target ipv4:192.168.6.200
codec g729br8
dial-peer voice 20 voip
description incoming from IP to SIP PBX
session protocol sipv2
incoming called-number 60..
dtmf-relay h245-alphanumeric
codec g729br8
Thanks for tips in advance.
10-16-2009 03:48 AM
Can you take "debug isdn q931" and "debug ccsip message" ? Do not enable other debugs.
10-16-2009 04:00 AM
I have finally brought my message to readable form. The isdn debug is in the message itself. Any ideas?
I will take sip debug and post it but I don't know how it will help as gw accounting probably takes info from sip messages, so they should show timing and codes to be expected but not ocurring..
10-16-2009 04:06 AM
debug ccsip message is needed to see what PBX is actually sending, it is not safe to infer things when troubleshooting.
Also metnion exact IOS version please.
10-16-2009 04:44 AM
here is the consolidated debug. please see the attachment as it doesn't fit in 4000 letters.
notice weird thing: user 350 calls 6008.
panasonic dials 600 and adds 8 as info.
later there are two gw acc records:
the last one goes for 350 to 600 and correlates with isdn timing... (shows gap in time and states normal clearing)
what is it I wonder?
10-16-2009 04:53 AM
Which exact IOS is this ?
(as asked before).
10-16-2009 04:53 AM
here are sw versions:
3825
Cisco IOS Software, 3800 Software (C3825-ADVIPSERVICESK9-M), Version 12.4(15)T9,
RELEASE SOFTWARE (fc5)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2009 by Cisco Systems, Inc.
Compiled Tue 28-Apr-09 17:45 by prod_rel_team
ROM: System Bootstrap, Version 12.4(13r)T11, RELEASE SOFTWARE (fc1)
2801: (sorry, it is 2801, not 2811 as I stated before)
Cisco IOS Software, 2801 Software (C2801-ADVIPSERVICESK9-M), Version 12.4(15)T7,
RELEASE SOFTWARE (fc3)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2008 by Cisco Systems, Inc.
Compiled Wed 13-Aug-08 18:11 by prod_rel_team
ROM: System Bootstrap, Version 12.4(13r)T, RELEASE SOFTWARE (fc1)
10-16-2009 05:00 AM
Try upgrading for example, 12.4(22)YB4 and check again.
10-16-2009 05:09 AM
that is not easily possible. one router is many miles away and have slow and jerky internet connectivity. and I don't see good reason for that: these images are quite good and stable. do you still think that it is not a configuration issue but a software bug?
10-16-2009 05:13 AM
Yes. Check "show sip-ua map sip-to-pstn" and you will see the mapping is correct, but does not happen.
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10-16-2009 05:20 AM
I've issued that command.
Here's the result:
Kievskaya#sh sip-ua map sip-pstn
The SIP Status code to PSTN Cause mapping table:-
SIP-Status Configured Default
PSTN-Cause PSTN-Cause
400 127 127
401 57 57
402 21 21
403 57 57
404 1 1
405 127 127
406 127 127
407 21 21
408 102 102
409 41 41
410 1 1
413 127 127
414 127 127
415 79 79
416 127 127
417 63 63
420 127 127
422 127 127
480 18 18
481 127 127
482 127 127
483 127 127
484 28 28
485 1 1
486 17 17
487 16 16
488 127 127
500 41 41
501 79 79
502 38 38
503 63 63
504 102 102
505 127 127
580 49 49
600 17 17
603 21 21
604 1 1
606 58 58
Don't know what you mean.
Anyway, no sip-ua is used. We use sip dial-peers without sip-server. Just peers. On sip-enabled panasonic in remote location with some help from the manufacturer the need to register on server was removed and so it also works in peer-to-peer configuration.
10-16-2009 07:00 AM
As the header says, means SIP cause 486 maps to ISDN cause 17. But in your cause that does not happen.
Even if you don't configure it, sip-ua is enabled anyway in the router. It-s just part of the whole SIP process.
10-16-2009 07:33 AM
I got it and I see SIP 486 busy in debug
but I don't get the point - in the very first post I've already said that gw acc shows good code so sip should do the same..
I'm worried about the "extra" gw accounting syslog message with false destination number in the last line. What can it relate too?
We can upgrade software in the hub node.. but it means getting bugs. Actually we had nat-related bugs in newer releases so this one was chosen for stable operation.
It could well be that solving this problem via sw upgrade will cause some more severe complications.
10-16-2009 07:42 AM
As you noted, IOS has many bugs, one can be the number is syslog, I did not paid attention to that.
You can only hope that at one point you find an image where everything works, or if you have a support contract, complain to the TAC.
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10-16-2009 12:39 PM
I searched and found a severity 2 bug. Ignore the fact is about H.323 and a different IOS, it may very well apply to you.
CSCtb99736
H323 disconnect cause is not fowarded to application
Symptom: ISDN Cause codes are not being forwarded transparently to the PBX.
Conditions: Issue was found with 7206VXR running 12.4(22)T1
Workaround: none
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