SIP "INVITE" problem on Cisco router 3945

babatunde_sanda · ‎03-29-2012

Hello Group. I have this unique problem. I have a SIP trunk setup to Broadworks. I have it configured using a Pilot TN for the trunk users which are basicly other TNs (Visualize it as Parent-child relationship). This trunk is configured to registerd to a Cisco 3945 router. It registers fine and I can make calls when I configure a dial-peer that matches exactly one of the registered TN on the trunk. This dial-peers actually register to the Broadworks system.

My problem is I cannot afford to configured close to 4k (four thousand) TN using dial peers on this router for the calls to be successful.

The invite sent to the router has the Pilot user TN on it and it looks like below:

INVITE sip:313303XXXX@167.165.85.117:5060 SIP/2.0

From: <sip:voip.lab.org>;tag=3d53a489-13c4-4f741cf3-e840fb89-4b1efaa6

To: "Tom Jones" <sip:31330510XX@167.165.85.117>

Call-ID: BW092915138290312250054523@10.2.2.11

CSeq: 1 INVITE

Via: SIP/2.0/UDP 167.165.85.97:5060;branch=z9hG4bK-1d3eac-4f741cf3-e840fb89-6ef37ce7

Supported: 100rel

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

Max-Forwards: 9

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

Contact: <sip:voip.lab.org:5060;maddr=167.165.85.97;transport=udp>

Content-Type: application/sdp

Content-Length: 278

v=0

o=BroadWorks 659070 1 IN IP4 167.165.85.97

s=-

c=IN IP4 167.165.85.97

t=0 0

a=sendrecv

m=audio 24384 RTP/AVP 9 0 8 18 127

a=rtpmap:9 G722/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:127 telephone-event/8000

a=fmtp:18 annexb=no

Is there a way on the Cisco router I can replace the Invite with the information in the To:? (Take this

sip:313303XXXX@167.165.85.117 and replace with

sip:31330510XX@167.165.85.117) by doing this I can configure dial-peers on the Router with wildcard *.$^ etc so the call is completed. The router routes the call using the information in this "invite" instead of the "To:".

Any hint or better approach from both the router's perspective or Broadworks appreciated.

Thank you.

Daniele Giordano · ‎03-30-2012

Do you have already try with CUBE and a sip-profile?

Daniele Giordano · ‎03-30-2012

Something like this:

voice class sip-profiles 1

request INVITE peer-header sip TO copy “sip:(.*)@” u01

request INVITE sip-header SIP-Req-URI modify “.*@(.*)” “INVITE sip:\u01@\1″

Regards.

babatunde_sanda · ‎03-30-2012

Hello Daniele,

Thanks for your reply. I knew about the sip profiles I just didn't know how to get the right config I needed. I am going to try that today. But a thought in my head is how can I use it to match the incoming. I believe this is for outgoing if I applied it to the dail-peer or globally. Dial-peer taking precedence.

The change needs to happen when coming into the router so that the router can then use this information to look for a valid dial-peer to send it out to.

Thanks

Daniele Giordano · ‎03-30-2012

You can configure the "voice-class sip-profiles" on the inbound dial-peer.

Regards.

babatunde_sanda · ‎03-30-2012

Ok. I"II try it in an hour and update you. This is the plan.

Let me try and illustrate the flow again.

Call comes in from provider with this header over an ethernet SIP trunk

########################################################

INVITE sip:313303XXXX@167.165.85.117:5060 SIP/2.0

From: ;tag=3d53a489-13c4-4f741cf3-e840fb89-4b1efaa6

To: "Tom Jones" <31330510XX>

Call-ID: BW092915138290312250054523@10.2.2.11

CSeq: 1 INVITE

Via: SIP/2.0/UDP 167.165.85.97:5060;branch=z9hG4bK-1d3eac-4f741cf3-e840fb89-6ef37ce7

Supported: 100rel

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

Max-Forwards: 9

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

Contact:

Content-Type: application/sdp

Content-Length: 278

########################################################

Using an incoming dial-peer that looks like this

dial-peer voice 1 voip

destination-pattern .T

progress_ind setup enable 3

session protocol sipv2

session target sip-server

incoming called-number .

voice-class codec 1

voice-class sip asserted-id pai

voice-class sip privacy-policy passthru

voice-class sip profiles 1 ==============> applied sip profile

dtmf-relay rtp-nte

################

Configured Profile

voice class sip-profiles 1

request INVITE peer-header sip TO copy “sip:(.*)@” u01

request INVITE sip-header SIP-Req-URI modify “.*@(.*)” “INVITE sip:\u01@\1″

#################

The called numbers reside on PRI's with dial-peers examples below

dial-peer voice 100 pots

destination-pattern 1....

progress_ind setup enable 3

incoming called-number .

direct-inward-dial

port 0/0/0:23

forward-digits all

dial-peer voice 700 pots

destination-pattern 7....

progress_ind setup enable 3

incoming called-number .

direct-inward-dial

port 0/0/0:23

no digit-strip

babatunde_sanda · ‎03-30-2012

Hello Daniele,

Tried your config, it did not work. Tried applying it glabally and on the dial peer. Thank you.

Daniele Giordano · ‎03-30-2012

Are you sure that incoming call match the dial-peer 1?

Can you add the output of "debug voip dialpeer" and "debug ccsip messages"?

Regards.

babatunde_sanda · ‎03-31-2012

Hello Daniele,

Sorry for the late response. Had to go handle other pressing issues. Here is a snippet from the configs. The plan is to send 5 digits through the trunk to the router. And for the router to route the calls over the PRI's and VOIP dial-peer. As mentioned above, the invite shows up as

INVITE sip:313303XXXX@167.165.85.117:5060 SIP/2.0

and the To is

To: "Tom Jones" <510XX>

or

To: "James Balls" <610XX>

or

To: "Sam Hall" <110XX>

or

To: "Hector Hernadez" <810XX>

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw

sip

bind control source-interface GigabitEthernet1/1

bind media source-interface GigabitEthernet1/1

header-passing

error-passthru

outbound-proxy ipv4:130.164.80.4

no update-callerid

midcall-signaling passthru

privacy-policy passthru

controller T1 0/1/1

framing esf

clock source internal

linecode b8zs

pri-group timeslots 1-24

interface GigabitEthernet1/1

description Trunk_Interface

ip address 130.164.80.4 255.255.255.0

interface Serial0/1/1:23

no ip address

encapsulation hdlc

isdn switch-type primary-ni

isdn timer T310 40000

isdn protocol-emulate network

isdn incoming-voice voice

no cdp enable

dial-peer voice 1 pots

destination-pattern 1....

progress_ind setup enable 3

incoming called-number .

direct-inward-dial

port 0/1/1:23

no digit-strip

!

dial-peer voice 5 pots

destination-pattern 5....

progress_ind setup enable 3

incoming called-number .

direct-inward-dial

port 0/1/1:23

no digit-strip

!

dial-peer voice 6 pots

destination-pattern 6....

progress_ind setup enable 3

incoming called-number .

direct-inward-dial

port 0/1/1:23

no digit-strip

dial-peer voice 2 voip

preference 1

destination-pattern 8….

session target ipv4:130.164.80.2

incoming called-number .

voice-class codec 1

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

dial-peer voice 3 voip

destination-pattern .T

session protocol sipv2

session target sip-server

session transport udp

incoming called-number .

voice-class codec 1

voice-class sip asserted-id pai

voice-class sip privacy-policy passthru

dtmf-relay rtp-nte

The "TO:" needs to be the numbers in the "invite" for the call to be successful. I"II be at the router console on Monday. Hopefully I"II be successful at making it work. Let me have your suggestions. Thank you.