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Incoming call to Analogue phone via FXS issue

Hi Everyone,

Can someone please share the knoweldge of what could be the cause of the incoming call via SIP trunk being failed to reach analogue phone which is connected to a FXS port on CUBE? I have a site with a cave location where there was an existing telephone cabling and have provided an analogue phone in there. However, the SIP trunk is terminated on an another site which shares DID range and I have attached configuration of both CUBE routers.

Main site with SIP trunk: wan1r001 (Cisco 2911-VSEC-CUBE with 1 FXO and FXS)

The site with an analogue line: yan1r001 (Cisco 2911-VSEC-CUBE with 1 FXO and FXS)

CUCM Version: CUCM 8.6.2

I have also done 'debug ccsip message' and the following is its output when an incoming call was tested to the line. The call flow for that test call would be PSTN -> SIP SP -> CUBE at wan1r001 -> CUCM 8.6 > CUBE at yan1r001 -> end-point?

wan1r001#

Oct 29 05:43:36.647: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:893037779@172.19.86.2:5060 SIP/2.0

Via: SIP/2.0/UDP 203.52.2.164:5060;branch=z9hG4bKtlnijt204o30h0pte1k1.1

From: <sip:893340333@203.52.2.164;user=phone;url-cookie=VNWEHA1-a61ngtns76ee3>;tag=538692716-1383025416599-

To: "893037779 893037779"<sip:893037779@myorg.com.au;url-cookie=VNWEHA1-n7eublr6digf0>

Call-ID: BW164336599291013317570105@10.83.154.147

CSeq: 22973644 INVITE

Contact: <sip:893340333@203.52.2.164:5060;url-cookie=VNWEHA1-1uu40pjrhm1a0;transport=udp>

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

Accept: application/media_control+xml,application/sdp,multipart/mixed

Supported:

Max-Forwards: 29

Content-Type: application/sdp

Content-Length: 416

v=0

o=BroadWorks 890728072 1 IN IP4 203.52.2.164

s=-

c=IN IP4 203.52.2.164

t=0 0

a=media-release:hngl5rp9pujvivh05pvumib1j8rpvg8o25736svbbeur62084c30

a=media-release-con-addr:d6v1k1hvmhvh1081o9j0

m=audio 18146 RTP/AVP 8 18 0 101

a=ptime:20

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=fmtp:18 annexb=no

a=silenceSupp:off - - - -

Oct 29 05:43:36.655: //209336/E3D681B6B855/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 203.52.2.164:5060;branch=z9hG4bKtlnijt204o30h0pte1k1.1

From: <sip:893340333@203.52.2.164;user=phone;url-cookie=VNWEHA1-a61ngtns76ee3>;tag=538692716-1383025416599-

To: "893037779 893037779"<sip:893037779@myorg.com.au;url-cookie=VNWEHA1-n7eublr6digf0>

Date: Tue, 29 Oct 2013 05:43:36 GMT

Call-ID: BW164336599291013317570105@10.83.154.147

CSeq: 22973644 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

Oct 29 05:43:36.655: //209337/E3D681B6B855/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:7779@10.6.130.11:5060 SIP/2.0

Via: SIP/2.0/UDP 172.19.86.2:5060;branch=z9hG4bK528D6186E

Remote-Party-ID: <sip:00893340333@172.19.86.2>;party=calling;screen=no;privacy=off

From: <sip:00893340333@as.nipt.telstra.com>;tag=3122EC0-1952

To: <sip:7779@10.6.130.11>

Date: Tue, 29 Oct 2013 05:43:36 GMT

Call-ID: E3D71DDE-3F9311E3-B85BDB9E-82954B39@172.19.86.2

Supported: timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 3822485942-1066602979-3092634526-2190822201

User-Agent: Cisco-SIPGateway/IOS-12.x

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

CSeq: 101 INVITE

Timestamp: 1383025416

Contact: <sip:00893340333@172.19.86.2:5060>

Expires: 300

Allow-Events: telephone-event

Max-Forwards: 28

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 303

v=0

o=CiscoSystemsSIP-GW-UserAgent 6895 6200 IN IP4 172.19.86.2

s=SIP Call

c=IN IP4 172.19.86.2

t=0 0

m=audio 28006 RTP/AVP 8 18 0 101

c=IN IP4 172.19.86.2

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

wan1r001#

Oct 29 05:43:36.659: //209337/E3D681B6B855/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.19.86.2:5060;branch=z9hG4bK528D6186E

From: <sip:00893340333@as.nipt.telstra.com>;tag=3122EC0-1952

To: <sip:7779@10.6.130.11>

Date: Tue, 29 Oct 2013 05:43:36 GMT

Call-ID: E3D71DDE-3F9311E3-B85BDB9E-82954B39@172.19.86.2

CSeq: 101 INVITE

Allow-Events: presence

Content-Length: 0

wan1r001#

Oct 29 05:43:46.651: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

CANCEL sip:893037779@172.19.86.2:5060 SIP/2.0

Via: SIP/2.0/UDP 203.52.2.164:5060;branch=z9hG4bKtlnijt204o30h0pte1k1.1

CSeq: 22973644 CANCEL

From: <sip:893340333@203.52.2.164;user=phone;url-cookie=VNWEHA1-a61ngtns76ee3>;tag=538692716-1383025416599-

To: "893037779 893037779"<sip:893037779@myorg.com.au;url-cookie=VNWEHA1-n7eublr6digf0>

Call-ID: BW164336599291013317570105@10.83.154.147

Max-Forwards: 29

Content-Length: 0

Oct 29 05:43:46.655: //209336/E3D681B6B855/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 203.52.2.164:5060;branch=z9hG4bKtlnijt204o30h0pte1k1.1

From: <sip:893340333@203.52.2.164;user=phone;url-cookie=VNWEHA1-a61ngtns76ee3>;tag=538692716-1383025416599-

To: "893037779 893037779"<sip:893037779@myorg.com.au;url-cookie=VNWEHA1-n7eublr6digf0>

Date: Tue, 29 Oct 2013 05:43:46 GMT

Call-ID: BW164336599291013317570105@10.83.154.147

CSeq: 22973644 CANCEL

Content-Length: 0

Oct 29 05:43:46.655: //209337/E3D681B6B855/SIP/Msg/ccsipDisplayMsg:

Sent:

CANCEL sip:7779@10.6.130.11:5060 SIP/2.0

Via: SIP/2.0/UDP 172.19.86.2:5060;branch=z9hG4bK528D6186E

From: <sip:00893340333@as.nipt.telstra.com>;tag=3122EC0-1952

To: <sip:7779@10.6.130.11>

Date: Tue, 29 Oct 2013 05:43:36 GMT

Call-ID: E3D71DDE-3F9311E3-B85BDB9E-82954B39@172.19.86.2

CSeq: 101 CANCEL

Max-Forwards: 70

Timestamp: 1383025426

Reason: Q.850;cause=16

Content-Length: 0

Oct 29 05:43:46.655: //209336/E3D681B6B855/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 487 Request Cancelled

Via: SIP/2.0/UDP 203.52.2.164:5060;branch=z9hG4bKtlnijt204o30h0pte1k1.1

From: <sip:893340333@203.52.2.164;user=phone;url-cookie=VNWEHA1-a61ngtns76ee3>;tag=538692716-1383025416599-

To: "893037779 893037779"<sip:893037779@myorg.com.au;url-cookie=VNWEHA1-n7eublr6digf0>;tag=31255CC-211E

Date: Tue, 29 Oct 2013 05:43:46 GMT

Call-ID: BW164336599291013317570105@10.83.154.147

CSeq: 22973644 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=16

Content-Length: 0

Oct 29 05:43:46.659: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:893037779@172.19.86.2:5060 SIP/2.0

Via: SIP/2.0/UDP 203.52.2.164:5060;branch=z9hG4bKtlnijt204o30h0pte1k1.1

CSeq: 22973644 ACK

From: <sip:893340333@203.52.2.164;user=phone;url-cookie=VNWEHA1-a61ngtns76ee3>;tag=538692716-1383025416599-

To: "893037779 893037779"<sip:893037779@myorg.com.au;url-cookie=VNWEHA1-n7eublr6digf0>;tag=31255CC-211E

Call-ID: BW164336599291013317570105@10.83.154.147

Max-Forwards: 29

Content-Length: 0

Oct 29 05:43:46.663: //209337/E3D681B6B855/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.19.86.2:5060;branch=z9hG4bK528D6186E

From: <sip:00893340333@as.nipt.telstra.com>;tag=3122EC0-1952

To: <sip:7779@10.6.130.11>

Date: Tue, 29 Oct 2013 05:43:46 GMT

Call-ID: E3D71DDE-3F9311E3-B85BDB9E-82954B39@172.19.86.2

CSeq: 101 CANCEL

Content-Length: 0

Oct 29 05:43:46.663: //209337/E3D681B6B855/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 487 Request Cancelled

Via: SIP/2.0/UDP 172.19.86.2:5060;branch=z9hG4bK528D6186E

From: <sip:00893340333@as.nipt.telstra.com>;tag=3122EC0-1952

To: <sip:7779@10.6.130.11>;tag=317832~57a3ab79-9f8e-4dee-acfc-ddcdab5c1646-53630368

Date: Tue, 29 Oct 2013 05:43:46 GMT

Call-ID: E3D71DDE-3F9311E3-B85BDB9E-82954B39@172.19.86.2

CSeq: 101 INVITE

Allow-Events: presence

Content-Length: 0

Oct 29 05:43:46.667: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

wan1r001#ACK sip:7779@10.6.130.11:5060 SIP/2.0

Via: SIP/2.0/UDP 172.19.86.2:5060;branch=z9hG4bK528D6186E

From: <sip:00893340333@as.nipt.telstra.com>;tag=3122EC0-1952

To: <sip:7779@10.6.130.11>;tag=317832~57a3ab79-9f8e-4dee-acfc-ddcdab5c1646-53630368

Date: Tue, 29 Oct 2013 05:43:36 GMT

Call-ID: E3D71DDE-3F9311E3-B85BDB9E-82954B39@172.19.86.2

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

wan1r001#

Thanks in advance for your assistance.

Regards,

1 ACCEPTED SOLUTION

Accepted Solutions

Incoming call to Analogue phone via FXS issue

layhlaing

You'll need to capture debugs simultaneously on both CUBE routers and capture an affected call:

PSTN -> SIP SP -> CUBE at wan1r001 -> CUCM 8.6 > CUBE at yan1r001 -> end-point?

First CUBE:

deb voice ccapi inout

deb ccsip messages

Second CUBE:

deb voice ccapi inout

deb ccsip messages

deb vpm signal

Please share the complete configuration for both routers.

HTH

--
Jorge Armijo

Please remember to rate helpful responses and identify helpful or correct answers.

-- Jorge Armijo Please remember to rate helpful responses and identify helpful or correct answers.
4 REPLIES

Incoming call to Analogue phone via FXS issue

layhlaing

You'll need to capture debugs simultaneously on both CUBE routers and capture an affected call:

PSTN -> SIP SP -> CUBE at wan1r001 -> CUCM 8.6 > CUBE at yan1r001 -> end-point?

First CUBE:

deb voice ccapi inout

deb ccsip messages

Second CUBE:

deb voice ccapi inout

deb ccsip messages

deb vpm signal

Please share the complete configuration for both routers.

HTH

--
Jorge Armijo

Please remember to rate helpful responses and identify helpful or correct answers.

-- Jorge Armijo Please remember to rate helpful responses and identify helpful or correct answers.
New Member

Re: Incoming call to Analogue phone via FXS issue

Thanks Jorge for your help. I have replicated this issue and gathered debugs as attached. Complete configs are also attached.

FXS analogue lines that we use in the first CUBE are working fine but not this one at second CUBE. And I am seeing the CANCEL request in the debug on CUBE1 and the calling is failing over to "SIP failover number" which is there at SP level for "Business continuity" which should only happens when SIP trunk is out of service.

Can you please explain a bit of what you can see in the debugs?

Cheers,

New Member

Re: Incoming call to Analogue phone via FXS issue

Hi Jorge,

I am just letting you know that the issue has been resolved. It was the change that we made on the CUBE2 router few days ago. It happened when Nortel PoE switches were replaced with Cisco ones and it slipped my mind to see the issue. After updating the SIP trunk's IP address in CUCM, both incoming calls and outbound calls to/from that analogue phone was working. The correct amount of debugging level as you advised has pointed me out this so thanks again.

Cheers,

Lay

Re: Incoming call to Analogue phone via FXS issue

Good to know

--
Jorge Armijo

Please remember to rate helpful responses and identify helpful or correct answers.

-- Jorge Armijo Please remember to rate helpful responses and identify helpful or correct answers.
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