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New Member

[AS5350] SIP to ISDN, no calling number field

Hi all

There's a problem with calls from SIP domain to ISDN, calling number field missed in q931 setup message

Call is ok, but ISDN subscribers dont receive any clid at all.

Network Scheme

Naumen CC ------ AS5350 ------- PBX

Debug

#debug ccsi mess

INVITE sip:10001369@x.x.x.x:5060;transport=udp SIP/2.0

Via:SIP/2.0/UDP y.y.y.y:5072;branch=z9hG4bK16fea8c013d032e00d17a0627

To:<sip:10001369@x.x.x.x:5060;transport=udp>

From:"9051234567"<sip:9051234567@y.y.y.y:5072>;tag=e00d17a0496

Call-ID:32e00d-17a-259@y.y.y.y:5072

CSeq:1 INVITE

Contact:<sip:y.y.y.y:5072>

User-Agent:NCC-6.0.14.1

Allow:ACK,BYE,CANCEL,INVITE,MESSAGE,INFO,PING,PRACK

Supported:early-session, 100rel

Accept:application/sdp,application/dtmf,application/dtmf-relay

Max-Forwards:70

Content-Type:application/sdp

Content-Length:296

#debug isdn q931

        Bearer Capability i = 0x8090A3

                Standard = CCITT

                Transer Capability = Speech

                Transfer Mode = Circuit

                Transfer Rate = 64 kbit/s

        Channel ID i = 0xA9838A

                Exclusive, Channel 10

        Called Party Number i = 0x81, '369'

                Plan:ISDN, Type:Unknown

Config

dial-peer voice 10002 pots

trunkgroup SI-2000

description #1_to_si200_isdn

preference 1

destination-pattern 10001T

direct-inward-dial

interface Serial3/1:15

no ip address

trunk-group SI-2000 1

isdn switch-type primary-net5

isdn incoming-voice modem 64

isdn T310 50000

isdn sending-complete

isdn outgoing-voice info-transfer-capability 3.1kHz-audio

isdn outgoing display-ie

isdn outgoing ie redirecting-number

fair-queue 64 256 0

!

interface Serial3/2:15

no ip address

trunk-group SI-2000 2

isdn switch-type primary-net5

isdn incoming-voice modem 64

isdn T310 50000

isdn sending-complete

isdn outgoing-voice info-transfer-capability 3.1kHz-audio

isdn outgoing display-ie

isdn outgoing ie redirecting-number

fair-queue 64 256 0

Have somebody any ideas?

1 ACCEPTED SOLUTION

Accepted Solutions

Re: [AS5350] SIP to ISDN, no calling number field

The problem is in the debug output:

Sep 17 11:09:17.226: ISDN Se3/1:15 EVENTd: pak_private_number: Calling Number IE is being stripped

What is the IOS version?

Upgrade the IOS.

Regards.

7 REPLIES

[AS5350] SIP to ISDN, no calling number field

Can you add a full debug?

Can Naumen CC add Remote Party ID or P-Asserted ID in SIP signalling?

Regards.

New Member

Re: [AS5350] SIP to ISDN, no calling number field

Thnx for response!

What do you mean by saying full debug, logging trap 7?

There are many simultaneous calls on this device, so i'm afraid a bit.

I may be mistaken but RPID and P-Asserted ID are replacing one another, arent they?

According a manual, Naumen Call center uses RPID.

Re: [AS5350] SIP to ISDN, no calling number field

Can you add all messages captured from debug? I want see the debug of entire call, if possible.

I don't see RPID in your INVITE:

INVITE sip:10001369@x.x.x.x:5060;transport=udp SIP/2.0

Via:SIP/2.0/UDP y.y.y.y:5072;branch=z9hG4bK16fea8c013d032e00d17a0627

To:<10001369>

From:"9051234567"<9051234567>;tag=e00d17a0496

Call-ID:32e00d-17a-259@y.y.y.y:5072

CSeq:1 INVITE

Contact:

User-Agent:NCC-6.0.14.1

Allow:ACK,BYE,CANCEL,INVITE,MESSAGE,INFO,PING,PRACK

Supported:early-session, 100rel

Accept:application/sdp,application/dtmf,application/dtmf-relay

Max-Forwards:70

Content-Type:application/sdp

Content-Length:296

Regards.

New Member

Re: [AS5350] SIP to ISDN, no calling number field

Field "From:" is present in all SIP messages that cisco receives.

I've just known that NCC ver 6. doesn't use nor RPID neither P-Asserted ID, only From to send caller information.

I called IDSN subscriber from testing asterisk PBX with sendrpid=yes

INVITE sip:10001369@192.168.12.2 SIP/2.0

Via: SIP/2.0/UDP 192.168.12.200:5060;branch=z9hG4bK7afee1a3

Max-Forwards: 70

From: <2122>;tag=as5e0343d5

To: <10001369>

Contact: <2122>

Call-ID: 2979de1055918c1c36019f983b0cbfb4@192.168.12.200:5060

CSeq: 102 INVITE

User-Agent: FPBX-2.8.1(11.4.0)

Date: Tue, 17 Sep 2013 14:24:31 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Remote-Party-ID: "2122" <2122>;party=calling;privacy=off;screen=no

Content-Type: application/sdp

Content-Length: 284

v=0

o=root 523498440 523498440 IN IP4 192.168.12.200

s=Asterisk PBX 11.4.0

c=IN IP4 192.168.12.200

t=0 0

m=audio 11292 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

Sep 17 10:54:31.178: Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.12.200:5060;branch=z9hG4bK7afee1a3

From: <2122>;tag=as5e0343d5

To: <10001369>;tag=FA33BA64-21E5

Date: Tue, 17 Sep 2013 10:54:31 GMT

Call-ID: 2979de1055918c1c36019f983b0cbfb4@192.168.12.200:5060

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 102 INVITE

Allow-Events: telephone-event

Content-Length: 0

Sep 17 10:54:31.182: ISDN Se3/1:15 Q931: TX -> SETUP pd = 8  callref = 0x69B8

        Sending Complete

        Bearer Capability i = 0x8090A3

                Standard = CCITT

                Transer Capability = Speech

                Transfer Mode = Circuit

                Transfer Rate = 64 kbit/s

        Channel ID i = 0xA98395

                Exclusive, Channel 21

        Display i = '2122'

        Called Party Number i = 0x81, '369'

                Plan:ISDN, Type:Unknown

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.12.200:5060;branch=z9hG4bK7afee1a3

From: <2122>;tag=as5e0343d5

To: <10001369>;tag=FA33BA64-21E5

Date: Tue, 17 Sep 2013 10:54:31 GMT

Call-ID: 2979de1055918c1c36019f983b0cbfb4@192.168.12.200:5060

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 102 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO

Allow-Events: telephone-event

Contact: <10001369>

Content-Type: application/sdp

Content-Length: 240

v=0

o=CiscoSystemsSIP-GW-UserAgent 6150 7340 IN IP4 192.168.12.2

s=SIP Call

c=IN IP4 192.168.12.2

t=0 0

m=audio 17628 RTP/AVP 0 100

c=IN IP4 192.168.12.2

a=rtpmap:0 PCMU/8000

a=rtpmap:100 X-NSE/8000

a=fmtp:100 192-194

a=ptime:20

Sep 17 10:54:34.290: Received:

ACK sip:10001369@192.168.12.2:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.12.200:5060;branch=z9hG4bK6d3532b0

Max-Forwards: 70

From: <2122>;tag=as5e0343d5

To: <10001369>;tag=FA33BA64-21E5

Contact: <2122>

Call-ID: 2979de1055918c1c36019f983b0cbfb4@192.168.12.200:5060

CSeq: 102 ACK

User-Agent: FPBX-2.8.1(11.4.0)

Content-Length: 0

I also turned on isdn outgoing display-ie on ports, connected to PBX, so you can see it in q391 setup, but calling nubmer isn't present still.

New Member

Re: [AS5350] SIP to ISDN, no calling number field

Also debug isdn standart output

Sep 17 11:09:16.822: ISDN Se3/1:15 Q931: TX -> CONNECT_ACK pd = 8  callref = 0x6A8B

Sep 17 11:09:17.226: ISDN Se3/1:15 EVENTd: process_pri_call: call id 0xACCF, number 369, speed 0, call type VOICE, redial No, CSM call No, pdata Yes

Sep 17 11:09:17.226: ISDN Se3/1:15 EVENTd: pak_private_number: Calling Number IE is being stripped

Sep 17 11:09:17.226: ISDN Se3/1:15 EVENTd: pak_private_number: Packing Calling Pty. Num. without digits

Sep 17 11:09:17.226: ISDN Se3/1:15 EVENTd: pak_private_number: called type/plan overridden by call_decode

Sep 17 11:09:17.226: ISDN Se3/1:15 Q931: isdn_parser_map_lkup: overriding plan/type for 369, unknown/unknown to isdn/unknown

Sep 17 11:09:17.226: ISDN Se3/1:15 SERROR: pak_private_number: didn't copy oct3a reason: not CALLER_NUMBER_IE

Sep 17 11:09:17.230: ISDN Se3/1:15 EVENTd: pak_public_number: Calling Number IE is being stripped

Sep 17 11:09:17.230: ISDN Se3/1:15 Q931: isdn_parser_map_lkup: overriding plan/type for 369, unknown/unknown to isdn/unknown

Sep 17 11:09:17.230: ISDN Se3/1:15 EVENTd: calltrkr_setup_received: isdn_info=1679514868l, call_id=0xACCF ORIGINATE

Sep 17 11:09:17.230: ISDN Se3/1:15 EVENTd: calltrkr_setup_received: calltracker disabled

Sep 17 11:09:17.230: ISDN Se3/1:15 BACKHAUL: srl_send_l3_pak:

        source_id = Q.931, dest_id = Q.921, prim = DL_DATA_REQ

        priv_len = 4 int_id = 0x641AD608 datasize = 44

Sep 17 11:09:17.230: ISDN Se3/1:15 BACKHAUL: data =  0x641AD60800000300024004CF00010800

Sep 17 11:09:17.230:    08026A8D05A104038090A31803A98388

Sep 17 11:09:17.230:    280432313232700481333639

Sep 17 11:09:17.230: ISDN Se3/1:15 Q931: TX -> SETUP pd = 8  callref = 0x6A8D

        Sending Complete

        Bearer Capability i = 0x8090A3

                Standard = CCITT

                Transer Capability = Speech

                Transfer Mode = Circuit

                Transfer Rate = 64 kbit/s

        Channel ID i = 0xA98388

                Exclusive, Channel 8

        Display i = '2122'

        Called Party Number i = 0x81, '369'

                Plan:ISDN, Type:Unknown


Re: [AS5350] SIP to ISDN, no calling number field

The problem is in the debug output:

Sep 17 11:09:17.226: ISDN Se3/1:15 EVENTd: pak_private_number: Calling Number IE is being stripped

What is the IOS version?

Upgrade the IOS.

Regards.

New Member

Re: [AS5350] SIP to ISDN, no calling number field

Thanks a lot for response, Daniele

unfortunately i cannot definitely say if it is smth wrong with IOS (it's truly old as time) because we need particular image to work with h323 Hughes Network devices, this AS5350 is production device and I havent more devices for tests.

But it seems you are right.

Regards, Eugene

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