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New Member

VoIP Gateway fail code in calls from SIP to PSTN

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
!

8 REPLIES

VoIP Gateway fail code in calls from SIP to PSTN

Hi David,

under 'voice service voip', could you please try configuring 'signaling forward rawmsg' and remove 'signaling forward none'?

//Suresh Please rate all the useful posts.
New Member

VoIP Gateway fail code in calls from SIP to PSTN

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.

VoIP Gateway fail code in calls from SIP to PSTN

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

//Suresh Please rate all the useful posts.
New Member

VoIP Gateway fail code in calls from SIP to PSTN

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" ;tag=as0c0cd221

To: <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" ;tag=as0c0cd221

To: <192.10.28.3>;tag=814B82A8-14C3

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" ;tag=as0767396d

To: <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" ;tag=as0767396d

To: <192.10.28.3>;tag=814B8304-1A91

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" ;tag=as11c295b0

To: <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" ;tag=as11c295b0

To: <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" ;tag=as11c295b0

To: <626666666>;tag=814BA274-5A8

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=off

Contact: <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" ;tag=as11c295b0

To: <626666666>;tag=814BA274-5A8

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=off

Contact: <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" ;tag=as11c295b0

To: <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" ;tag=as11c295b0

To: <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" ;tag=as11c295b0

To: <626666666>;tag=814BA274-5A8

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" ;tag=as11c295b0

To: <626666666>;tag=814BA274-5A8

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

VoIP Gateway fail code in calls from SIP to PSTN

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??

//Suresh Please rate all the useful posts.

VoIP Gateway fail code in calls from SIP to PSTN

Could you please try "cause-code legacy" under voice service voip n see if that helps??

//Suresh Please rate all the useful posts.
New Member

VoIP Gateway fail code in calls from SIP to PSTN

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

VoIP Gateway fail code in calls from SIP to PSTN

Thanks for the update David. Good to know the issue is fixed.

//Suresh Please rate all the useful posts.
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