Solved: SIP port for outbound messages

espinedo1 · ‎01-14-2014

Hello,

I have configured an 2901 cisco router as a VoIP gateway, and when calls from ISDN to IP network arrives I can see that the SIP messages arrives from a port different from 5060 and differs from one call to another.

It's possible to set this port somethin fix?

For example: I have the cisco router connected to ISDN and to an Asterisk PBX, when a call from ISDN to Asterisk arrives, the INVITE message that arrives to Asterisk is as follows:

<--- SIP read from 89.1.28.3:58178 ---> ---->I'd like this port was something fix
INVITE sip:XXXXXXXXXX@89.1.23.201:5060 SIP/2.0
Via: SIP/2.0/UDP 89.1.28.3:5060;branch=z9hG4bK9C231F
Remote-Party-ID: <sip:YYYYYYYYY@89.1.28.3>;party=calling;screen=yes;privacy=off
From: <sip:YYYYYYYYY@89.1.28.3>;tag=24F5B7F8-935
To: <sip:XXXXXXXXXX@89.1.23.201>

Any idea?

bernhardczapp · ‎01-16-2014

To use the listener port for sending requests over the User Datagram Protocol (UDP)

it is:

sip-ua

connection-reuse

Have a look here:

http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_s11.html#wp1135436

View solution in original post

Suresh Subramanian · ‎01-14-2014

In the gateway, you can configure the port number with session target ipv4:89.X.X.X:5060 towards pbx.

//Suresh Please rate all the useful posts.

espinedo1 · ‎01-14-2014

Yes, the SIP messages are sent to the 5060 PBX port (the defaul port) .

But are sent from a dinamic port from the Cisco gateway (58178 in the example): The port I'd like to fix is the port of the gateway from which the invite is sent to the PBX

Suresh Subramanian · ‎01-16-2014

Hello David,

could you please capture the debug ccsip message from the GW for a call that has the port number different from 5060?

we need to capture the logs from pbx side as well for the same call.

also could you please post your GW config?

Please rate all the useful posts

//Suresh Please rate all the useful posts.

espinedo1 · ‎01-16-2014

Thanks sureshsub2 for your interest,

I attach the suggested captures.

As you can see in the PBX_SIP_capture --> The SIP invite received in the Asterisk PBX comes from port 55625 of the Cisco gateway (89.1.28.3)

Suresh Subramanian · ‎01-16-2014

Hello David, I checked the logs from both GW & PBX.

>> INVITE sent from GW to PBX (GW logs)

Jan 16 08:48:31.041: //17431/CF3D07B78042/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:XXXXXXXXX@89.1.23.205:5060 SIP/2.0 <<< INVITE sent to PBX on the port 5060 >>>

Via: SIP/2.0/UDP 89.1.28.3:5060;branch=z9hG4bKED10E9 <>

Remote-Party-ID: ;party=calling;screen=yes;privacy=off

From: ;tag=2DA91FAC-B7C

To:

Date: Thu, 16 Jan 2014 08:48:31 GMT

Call-ID: CF3DA3DF-7DC111E3-87C9A612-72F24CC1@89.1.28.3

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

Cisco-Guid: 3476883383-2109805027-2151833715-1547148384

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

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

CSeq: 102 INVITE

Max-Forwards: 70

Timestamp: 1389862111

Contact:

Expires: 180

Allow-Events: telephone-event

Authorization: Digest username="cisco_user",realm="asterisk",uri="sip:XXXXXXXXX@89.1.23.205:5060",response="924bbb0dee2d473ad080501eccc704b6",nonce="0ba17dd6",algorithm=MD5

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 426

v=0

o=CiscoSystemsSIP-GW-UserAgent 9375 3076 IN IP4 89.1.28.3 << Originator of the SIP message, the GW>>

s=SIP Call

c=IN IP4 89.1.28.3

t=0 0

m=audio 18814 RTP/AVP 98 99 9 15 18 8 0 101

c=IN IP4 89.1.28.3

a=rtpmap:98 G726-32/8000

a=rtpmap:99 G726-16/8000

a=rtpmap:9 G722/8000

a=fmtp:9 bitrate=64

a=rtpmap:15 G728/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

>> Here you see the same SIP INVITE message we received from GW in the PBX logs and it uses the same port 5060.

#

U 89.1.28.3:55625 -> 89.1.23.205:5060 ---> It seems,This line is something PBX added & you can see the same line at the top of every SIP message we receive from GW to PBX.

INVITE sip:XXXXXXXXX@89.1.23.205:5060 SIP/2.0.

Via: SIP/2.0/UDP 89.1.28.3:5060;branch=z9hG4bKED10E9.

Remote-Party-ID: ;party=calling;screen=yes;privacy=off.

From: ;tag=2DA91FAC-B7C.

To: .

Date: Thu, 16 Jan 2014 08:48:31 GMT.

Call-ID: CF3DA3DF-7DC111E3-87C9A612-72F24CC1@89.1.28.3.

Supported: 100rel,timer,resource-priority,replaces,sdp-anat.

Min-SE: 1800.

Cisco-Guid: 3476883383-2109805027-2151833715-1547148384.

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

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

CSeq: 102 INVITE.

Max-Forwards: 70.

Timestamp: 1389862111.

Contact: . <<Where the CallingParty can be reached for the return signaling path>>

Expires: 180.

Allow-Events: telephone-event.

Authorization: Digest username="cisco_user",realm="asterisk",uri="sip:XXXXXXXXX@89.1.23.205:5060",response="924bbb0dee2d473ad080501eccc704b6",nonce="0ba17dd6",algorithm=MD5.

Content-Type: application/sdp.

Content-Disposition: session;handling=required.

Content-Length: 426.

.

v=0.

o=CiscoSystemsSIP-GW-UserAgent 9375 3076 IN IP4 89.1.28.3.

s=SIP Call.

c=IN IP4 89.1.28.3.

t=0 0.

m=audio 18814 RTP/AVP 98 99 9 15 18 8 0 101.

c=IN IP4 89.1.28.3.

a=rtpmap:98 G726-32/8000.

a=rtpmap:99 G726-16/8000.

a=rtpmap:9 G722/8000.

a=fmtp:9 bitrate=64.

a=rtpmap:15 G728/8000.

a=rtpmap:18 G729/8000.

a=fmtp:18 annexb=no.

a=rtpmap:8 PCMA/8000.

a=rtpmap:0 PCMU/8000.

a=rtpmap:101 telephone-event/8000.

a=fmtp:101 0-

>> Not sure why PBX is adding that line for the sip messages we receive from GW to PBX.

>> However when the PBX responds to GW SIP messages, it is adding the line with correct port.

#

U 89.1.23.205:5060 -> 89.1.28.3:5060

SIP/2.0 100 Trying.

Via: SIP/2.0/UDP 89.1.28.3:5060;branch=z9hG4bKED10E9;received=89.1.28.3.

From: ;tag=2DA91FAC-B7C.

To: .

Call-ID: CF3DA3DF-7DC111E3-87C9A612-72F24CC1@89.1.28.3.

CSeq: 102 INVITE.

Server: Asterisk PBX 11.2.1.

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

Supported: replaces, timer.

Contact: .

Content-Length: 0.

#

U 89.1.23.205:5060 -> 89.1.28.3:5060

SIP/2.0 200 OK.

Via: SIP/2.0/UDP 89.1.28.3:5060;branch=z9hG4bKED10E9;received=89.1.28.3.

From: ;tag=2DA91FAC-B7C.

To: ;tag=as14ab9f2b.

Call-ID: CF3DA3DF-7DC111E3-87C9A612-72F24CC1@89.1.28.3.

CSeq: 102 INVITE.

Server: Asterisk PBX 11.2.1.

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

Supported: replaces, timer.

Contact: . << Where the called party can be reached for the signaling path >>

Content-Type: application/sdp.

Content-Length: 382.

with these captures, I would say, the SIP messages are exchanged between PBX & GW with port only 5060 with correct signaling IP addresses.

And the line we see in the PBX log (U 89.1.28.3:55625 -> 89.1.23.205:5060) is something added by the PBX not by the GW.

Though I don't see any significance for the line in the SIP signalling, The PBX technicians should be able to say why the line is added in the logs.

Please let me know if you have any questions on this.

Thanks

Suresh

Please rate all the useful posts

//Suresh Please rate all the useful posts.

bernhardczapp · ‎01-16-2014

To use the listener port for sending requests over the User Datagram Protocol (UDP)

it is:

sip-ua

connection-reuse

Have a look here:

http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_s11.html#wp1135436

espinedo1 · ‎01-16-2014

Thankyou bernhadczapp!!

That was what I was loking for