03-29-2012 03:44 PM
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.
03-30-2012 12:16 AM
Do you have already try with CUBE and a sip-profile?
03-30-2012 12:21 AM
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.
03-30-2012 05:22 AM
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
03-30-2012 07:32 AM
You can configure the "voice-class sip-profiles" on the inbound dial-peer.
Regards.
03-30-2012 08:06 AM
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:
To: "Tom Jones" <31330510XX>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
03-30-2012 10:29 AM
Hello Daniele,
Tried your config, it did not work. Tried applying it glabally and on the dial peer. Thank you.
03-30-2012 11:31 AM
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.
03-31-2012 04:51 PM
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>510XX>
or
To: "James Balls" <610XX>610XX>
or
To: "Sam Hall" <110XX>110XX>
or
To: "Hector Hernadez" <810XX>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.
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