12-30-2013 09:05 AM - edited 03-16-2019 09:02 PM
Hello,
I have configured a Cisco 2901 router with VWIC3 card as VoIP gateway. But I'm having the following problem:
When I throw calls from de IP network (with Asterisk PBX) to the PSTN network, and the call is rejected by the end user, the call end never arrives to my Asterisk PBX. I can see in logs that the busy code arrives from PSTN to the 2901, but no SIP message is sent to the Asterisk PBX, so it can not know wich was the fail cause of the call.
Have I to set any additional option in "dial-peer" or in "voice service voip"?
The Asterisk PBX is connected to GigabitEthernet0/0 and the PSTN connection is an E1 primary trunk
My config file is:
isdn switch-type primary-net5
isdn voice-call-failure 500
voice service voip
signaling forward none
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
!
!
interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
no cdp enable
!
!
dial-peer voice 1 pots
description inbound calls from PSTN (inbound leg)
incoming called-number 936019999
direct-inward-dial
!
dial-peer voice 2 voip
description outbound calls from Asterisk (inbound leg)
session protocol sipv2
incoming called-number .
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 3 voip
description inbound calls to Asterisk (outbound leg)
destination-pattern 936019999
session protocol sipv2
session target ipv4:89.1.23.205
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 4 pots
description outbound calls from Asterisk (outbound leg)
destination-pattern .
port 0/0/0:15
!
12-30-2013 09:54 AM
Hi David,
under 'voice service voip', could you please try configuring 'signaling forward rawmsg' and remove 'signaling forward none'?
12-30-2013 10:14 AM
Hi sureshsub2
I have tryied it (and also "unconditional"), but the SIP message with the fail reason is never generated. The "signaling forward rawmsg/unconditional" adds information in the SIP messages payload (183, session progress), but doesn't generate the SIP message, with the fail cause for Asterisk, when the end user rejects the call.
12-30-2013 10:22 AM
the cause code mapping between the pstn and sip is set by default. not sure why it is not generating the cause code in SIP leg. could you please provide the complete 'debug ccsip message & debug voip ccapi inout & debug isdn q931' for a test call
12-30-2013 10:48 AM
As you can see after the
Cause i = 0x829100000000 - User busy
arrives. The Cisco router doesn't send any SIP message. Only when the timeout for processing the call expires the Asterisk is who sends a CANCEL SIP message
Dec 30 18:29:54.266: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:192.10.28.3 SIP/2.0
Via: SIP/2.0/UDP 192.10.23.205:5060;branch=z9hG4bK2c35599d
Max-Forwards: 70
From: "asterisk"
To: <192.10.28.3>192.10.28.3>
Contact:
Call-ID: 48ed11e60c33e0c65eaaa6bf4c0080ab@192.10.23.205:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.2.1
Date: Mon, 30 Dec 2013 18:39:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
Dec 30 18:29:54.266: //936/363EE66984D0/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.10.23.205:5060;branch=z9hG4bK2c35599d
From: "asterisk"
To: <192.10.28.3>;tag=814B82A8-14C3192.10.28.3>
Date: Mon, 30 Dec 2013 18:29:54 GMT
Call-ID: 48ed11e60c33e0c65eaaa6bf4c0080ab@192.10.23.205:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 437
v=0
o=CiscoSystemsSIP-GW-UserAgent 421 3247 IN IP4 192.10.28.3
s=SIP Call
c=IN IP4 192.10.28.3
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 192.10.28.3
m=image 0 udptl t38
c=IN IP4 192.10.28.3
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy
Dec 30 18:29:54.358: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:192.10.28.3 SIP/2.0
Via: SIP/2.0/UDP 192.10.23.205:5060;branch=z9hG4bK5582f825
Max-Forwards: 70
From: "asterisk"
To: <192.10.28.3>192.10.28.3>
Contact:
Call-ID: 45aef0dc39dbbae8048cee446b4f80db@192.10.23.205:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.2.1
Date: Mon, 30 Dec 2013 18:39:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
Dec 30 18:29:54.362: //937/364CEFCA84D1/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.10.23.205:5060;branch=z9hG4bK5582f825
From: "asterisk"
To: <192.10.28.3>;tag=814B8304-1A91192.10.28.3>
Date: Mon, 30 Dec 2013 18:29:54 GMT
Call-ID: 45aef0dc39dbbae8048cee446b4f80db@192.10.23.205:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 437
v=0
o=CiscoSystemsSIP-GW-UserAgent 334 4649 IN IP4 192.10.28.3
s=SIP Call
c=IN IP4 192.10.28.3
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 192.10.28.3
m=image 0 udptl t38
c=IN IP4 192.10.28.3
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy
Dec 30 18:30:02.294: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:626666666@192.10.28.3 SIP/2.0
Via: SIP/2.0/UDP 192.10.23.205:5060;branch=z9hG4bK039d95dd
Max-Forwards: 70
From: "Anonymous"
To: <626666666>626666666>
Contact:
Call-ID: 312213da7a749a831659a66c35662942@192.10.23.205:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.2.1
Date: Mon, 30 Dec 2013 18:39:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 303
v=0
o=root 55449278 55449278 IN IP4 192.10.23.205
s=Asterisk PBX 11.2.1
c=IN IP4 192.10.23.205
t=0 0
m=audio 10678 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Dec 30 18:30:02.298: //938/3B07E01A84D2/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.10.23.205:5060;branch=z9hG4bK039d95dd
From: "Anonymous"
To: <626666666>626666666>
Date: Mon, 30 Dec 2013 18:30:02 GMT
Call-ID: 312213da7a749a831659a66c35662942@192.10.23.205:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Dec 30 18:30:02.302: ISDN Se0/0/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x1, Calling num anonymous
Dec 30 18:30:02.302: ISDN Se0/0/0:15 Q931: Sending SETUP callref = 0x00A7 callID = 0x8028 switch = primary-net5 interface = User
Dec 30 18:30:02.302: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8 callref = 0x00A7
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9838A
Exclusive, Channel 10
Calling Party Number i = 0x01A0, 'anonymous'
Plan:ISDN, Type:Unknown
Called Party Number i = 0x81, '626666666'
Plan:ISDN, Type:Unknown
Dec 30 18:30:02.394: ISDN Se0/0/0:15 Q931: RX <- SETUP_ACK pd = 8 callref = 0x80A7
Channel ID i = 0xA9838A
Exclusive, Channel 10
Dec 30 18:30:02.406: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x80A7
Progress Ind i = 0x8288 - In-band info or appropriate now available
Dec 30 18:30:02.406: //-1/xxxxxxxxxxxx/SIP/Msg/sipDisplayBinaryData:
Sending: Binary Message Body
Dec 30 18:30:02.406: Content-Type: application/x-q931
08 02 00 A7 02 1E 02 82 88 0D 0A
Dec 30 18:30:02.410: //938/3B07E01A84D2/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.10.23.205:5060;branch=z9hG4bK039d95dd
From: "Anonymous"
To: <626666666>;tag=814BA274-5A8626666666>
Date: Mon, 30 Dec 2013 18:30:02 GMT
Call-ID: 312213da7a749a831659a66c35662942@192.10.23.205:5060
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <626666666>;party=called;screen=no;privacy=off626666666>
Contact: <626666666>626666666>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: multipart/mixed;boundary=uniqueBoundary
Mime-Version: 1.0
Content-Length: 493
--uniqueBoundary
Content-Type: application/sdp
Content-Disposition: session;handling=required
v=0
o=CiscoSystemsSIP-GW-UserAgent 5199 9794 IN IP4 192.10.28.3
s=SIP Call
c=IN IP4 192.10.28.3
t=0 0
m=audio 18614 RTP/AVP 0 101
c=IN IP4 192.10.28.3
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
--uniqueBoundary
Content-Type: application/x-q931
Content-Disposition: signal;handling=optional
Content-Length: 11
.'
--uniqueBoundary--
Dec 30 18:30:04.814: %CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on GigabitEthernet0/0 (not half duplex), with SEP0004f2b01793 Port 1 (half duplex).
Dec 30 18:30:07.642: ISDN Se0/0/0:15 Q931: RX <- ALERTING pd = 8 callref = 0x80A7
Progress Ind i = 0x8288 - In-band info or appropriate now available
Dec 30 18:30:07.642: //-1/xxxxxxxxxxxx/SIP/Msg/sipDisplayBinaryData:
Sending: Binary Message Body
Dec 30 18:30:07.642: Content-Type: application/x-q931
08 02 00 A7 01 1E 02 82 88 0D 0A
Dec 30 18:30:07.646: //938/3B07E01A84D2/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.10.23.205:5060;branch=z9hG4bK039d95dd
From: "Anonymous"
To: <626666666>;tag=814BA274-5A8626666666>
Date: Mon, 30 Dec 2013 18:30:02 GMT
Call-ID: 312213da7a749a831659a66c35662942@192.10.23.205:5060
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <626666666>;party=called;screen=no;privacy=off626666666>
Contact: <626666666>626666666>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: multipart/mixed;boundary=uniqueBoundary
Mime-Version: 1.0
Content-Length: 493
--uniqueBoundary
Content-Type: application/sdp
Content-Disposition: session;handling=required
v=0
o=CiscoSystemsSIP-GW-UserAgent 5199 9794 IN IP4 192.10.28.3
s=SIP Call
c=IN IP4 192.10.28.3
t=0 0
m=audio 18614 RTP/AVP 0 101
c=IN IP4 192.10.28.3
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
--uniqueBoundary
Content-Type: application/x-q931
Content-Disposition: signal;handling=optional
Content-Length: 11
.'
--uniqueBoundary--
Dec 30 18:30:12.834: ISDN Se0/0/0:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x80A7
Cause i = 0x829100000000 - User busy
Progress Ind i = 0x8288 - In-band info or appropriate now available
Dec 30 18:30:12.834: ISDN Se0/0/0:15 Q931: call_disc: PI received in disconnect; Postpone sending RELEASE for callid 0x8028
Dec 30 18:30:22.294: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
CANCEL sip:626666666@192.10.28.3 SIP/2.0
Via: SIP/2.0/UDP 192.10.23.205:5060;branch=z9hG4bK039d95dd
Max-Forwards: 70
From: "Anonymous"
To: <626666666>626666666>
Call-ID: 312213da7a749a831659a66c35662942@192.10.23.205:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 11.2.1
Content-Length: 0
Dec 30 18:30:22.298: //938/3B07E01A84D2/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.10.23.205:5060;branch=z9hG4bK039d95dd
From: "Anonymous"
To: <626666666>626666666>
Date: Mon, 30 Dec 2013 18:30:22 GMT
Call-ID: 312213da7a749a831659a66c35662942@192.10.23.205:5060
CSeq: 102 CANCEL
Content-Length: 0
Dec 30 18:30:22.306: //938/3B07E01A84D2/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.10.23.205:5060;branch=z9hG4bK039d95dd
From: "Anonymous"
To: <626666666>;tag=814BA274-5A8626666666>
Date: Mon, 30 Dec 2013 18:30:22 GMT
Call-ID: 312213da7a749a831659a66c35662942@192.10.23.205:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=16
Content-Length: 0
Dec 30 18:30:22.310: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:626666666@192.10.28.3:5060 SIP/2.0
Via: SIP/2.0/UDP 192.10.23.205:5060;branch=z9hG4bK039d95dd
Max-Forwards: 70
From: "Anonymous"
To: <626666666>;tag=814BA274-5A8626666666>
Contact:
Call-ID: 312213da7a749a831659a66c35662942@192.10.23.205:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.2.1
Content-Length: 0
Dec 30 18:30:22.318: ISDN Se0/0/0:15 Q931: TX -> RELEASE pd = 8 callref = 0x00A7
Dec 30 18:30:22.594: ISDN Se0/0/0:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x80A7
12-30-2013 11:14 AM
Perhaps, It might be issue with the IOS. What is the IOS version??
"show sip-ua map pstn-sip" should display the correct mapping. Could you please check that once??
12-30-2013 11:32 AM
Could you please try "cause-code legacy" under voice service voip n see if that helps??
01-02-2014 04:00 AM
The problem seems to be that the CISCO postpone the forward of the call ends, as you can see in debug messages:
ISDN Se0/0/0:15 Q931: call_disc: PI received in disconnect; Postpone sending RELEASE for callid 0x8028
To solve it I have added
voice call disc-pi-off
in global configuration
With this option enabled the Cisco router sends the SIP message
SIP/2.0 486 Busy here.
when the disconnects arrives from the ISDN interface
01-02-2014 07:02 AM
Thanks for the update David. Good to know the issue is fixed.
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