06-17-2016 12:01 AM - edited 03-17-2019 07:16 AM
Hi,
I just have a problem to modify the "P-Asserted-Identity" as the provider expect a special value.
The "normal" unmodified P-Asserted-Identity looks like this
P-Asserted-Identity: "username" <sip:XXXX77@x.x.x.x>
I want to change this P-Asserted-Identity to this one
P-Asserted-Identity: "username"<sip:YYYYYYYY77@y.y.y.y:5060>
X = internal DN
x = internal IP of CUCM
Y = external DN
y = external IP
I tried to do this with this config
voice class sip-profiles 1
request ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*)<sip:XXXX77@x.x.x.x>" "P-Asserted-Identity:\1<sip:YYYYYYYY77@y.y.y.y:5060>"
dial-peer voice 11 voip
description SIP-Outgoing
translation-profile incoming SipInbound
translation-profile outgoing SipOutbound
destination-pattern 0T
modem passthrough nse codec g711alaw
session protocol sipv2
session target ipv4:y.y.y.y
session transport udp
voice-class sip early-offer forced
voice-class sip profiles 1
dtmf-relay sip-notify rtp-nte
codec g711alaw
fax-relay ecm disable
fax-relay sg3-to-g3
fax nsf 000000
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
no vad
!
In the log I also see that dialpeer 11 should be used:
*Jun 16 15:16:31.242 MESZ: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Outgoing Dial-peer=11 Is Matched
But the problem is that the P-Asserted-Identity is not modified and still the default value.
What did I wrong?
BR
Michael
Solved! Go to Solution.
06-21-2016 03:59 AM
ok, great
If the answers were useful please mark them as correct so we can consider this subject closed.
Leszek
06-17-2016 04:09 AM
Sometimes the debug dialpeer output is challenging to read. Did you confirm that the call was actually connected using PID:11 on the outbound call leg using show call active voice brief?
Assuming that's a yes, I agee with Leszek that it looks good syntactically; however, you did specify the request. You may want to add a second line to the profile for response.
voice class sip-profiles 1
request ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*)<sip:XXXX77@x.x.x.x>" "P-Asserted-Identity:\1<sip:YYYYYYYY77@y.y.y.y:5060>"
response ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*)<sip:XXXX77@x.x.x.x>" "P-Asserted-Identity:\1<sip:YYYYYYYY77@y.y.y.y:5060>"
For example, the 1XX messages (e.g. 180 Ringing or 183 Session Progress) are responses, not a request. Same story for the 200 OK and ACK messages, all of which are the most common places for PAI to be communicated by CUCM (1XX is alerting while 200 is connected party information).
06-17-2016 11:49 AM
Hi Michael,
I've been testing your script using this tool:
http://sip-profile.54.227.241.219.xip.io/
And for me all looks good (see the screenshot):
It might be a good idea to run "debug ccsip all" as this should debug also the script part and we should see why it's not applied.
Leszek
06-17-2016 11:49 AM
Hi Leszek,
Hi Jonathan,
thanks for your answers. I will have a look at this and reply to you.
@Leszek: I tried to open you mentioned test tool at "http://sip-profile.54.227.241.219.xip.io/sip-profiles.cgi" but the link is not working for me.
BR
Michael
06-17-2016 11:53 AM
Hi Jonathan,
the output for show call active voice brief looks like this:
Total call-legs: 2
0 : 12503 125141090ms.1 (*20:47:49.405 MESZ Fri Jun 17 2016) +23730 pid:100 Answer XXXX77 active
dur 00:00:04 tx:67/10100 rx:725/116000 dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 10.40.40.41:31106 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711alaw TextRelay: off Transcoded: No ICE: Off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
LostPacketRate:0.00 OutOfOrderRate:0.00
0 : 12504 125141100ms.1 (*20:47:49.415 MESZ Fri Jun 17 2016) +23710 pid:11 Originate EXTERNAL_PHONE_NUMBER active
dur 00:00:04 tx:725/116000 rx:109/16510 dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 10.15.165.102:17022 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711alaw TextRelay: off Transcoded: No ICE: Off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
LostPacketRate:0.00 OutOfOrderRate:0.00
BR
Michael
06-17-2016 11:58 AM
Michael,
Here is the correct link:
http://sip-profile.54.227.241.219.xip.io/
I've also updated in the previous post.
Leszek
06-17-2016 01:28 PM
Hi Leszek,
thank you.
In the test tool everything looks fine when I input:
SIP-Profile:
request ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*)<sip:XXXX77@x.x.x.x>" "P-Asserted-Identity:\1<sip:YYYYYYY77@y.y.y.y:5060>"
Input Message:
INVITE sip:0049123456789@a.b.c.d:5060 SIP/2.0
Via: SIP/2.0/TCP x.x.x.x:5060;branch=z9hG4bK586c875fd7d71
From: "CANCOM" <sip:XXXX77@x.x.x.x>;tag=1007421~bf696554-d8b2-48cf-bd09-1508ef59b52b-58497736
To: <sip:0049123456789@a.b.c.d>
Date: Fri, 17 Jun 2016 18:43:04 GMT
Call-ID: 51ef7180-764144b8-519c9-2928280a@10.40.40.41
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM10.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:x.x.x.x:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 1374646656-0000065536-0000000240-0690497546
Session-Expires: 1800
P-Asserted-Identity: "username" <sip:XXXX77@x.x.x.x>
Remote-Party-ID: "username" <sip:XXXX77@x.x.x.x>;party=calling;screen=yes;privacy=off
Contact: <sip:XXXX77@x.x.x.x:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 201
Output Message
P-Asserted-Identity: "username" <sip:YYYYYYYY77@y.y.y.y:5060>
But I don`t know why it`s not working at the router when calling.
BR
Michael
06-20-2016 02:06 AM
Michael,
You can collect "debug ccsip all" those debugs should capture the information of why the profile is not applied.
Leszek
06-20-2016 02:41 AM
Hi Leszek,
deb ccsip all is "not working" for that router because when I enable this debug I can`t manage the router anymore because the ping will increase up to over 2000 ms.
Is there any other ccsip debug like ccsip calls / messages ... to see the fault?
BR
Michael
06-20-2016 03:23 AM
Try following this dog for debug on the GW:
https://supportforums.cisco.com/document/62906/how-properly-and-safely-collect-debugs-ios-router
Leszek
06-20-2016 05:02 AM
06-20-2016 06:06 AM
Michael,
From what I can see PAI is not send on the outoging dial-peer to ITSP. Can you please add the command:
voice-class sip asserted-id pai
On the outgoing dial-peer. If it still doesn't work, please attach same set of traces with the command applied.
Leszek
06-20-2016 06:34 AM
Hi Leszek,
after adding this command to the outgoing dial-peer no outgoing calls are working anymore although I resetted the SIP trunk.
Could this be maybe because the provider disabled the "P-Asserted-Identity" query so we can call outside without the correct "P-Asserted-Identity"?
The "P-Asserted-Identity" also was not changed after adding this command:
P-Asserted-Identity: "username" <sip:XXXX77@x.x.x.x>
In the attachment the log from an unsuccesfull outgoing call.
BR
Michael
06-20-2016 07:07 AM
This time it was actually send. You can see different PAI is received and different is send. But ITSP is rejecting call because of invalid PAI. Not sure why is that they'd need to tell why do they do it. Would be good to check, I think it might be because of tel instead of sip. But would be good to check with them so we can modify it in the right way.
*Jun 20 15:22:40.470 MESZ: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0049123456789@a.b.c.c:5060 SIP/2.0
Session-Expires: 1800
P-Asserted-Identity: "username" <sip:XXXX77@x.x.x.x>
*Jun 20 15:22:40.486 MESZ: //36351/610836800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:0049123456789@y.y.y.y:5060 SIP/2.0
[...]
Max-Forwards: 69
P-Asserted-Identity: "username" <tel:YYYYYYYY77@a.b.c.c>
[...]
SIP: (36351) Group (a= group line) attribute, level 65535 instance 1 not found.
*Jun 20 15:22:41.170 MESZ: //36351/610836800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 400 Invalid P-Asserted-Identity
Leszek
06-20-2016 07:07 AM
Hi Leszek,
many thanks for explanation.
You`re right.
They already told me how the P-Asserted-Identity should look like:
P-asserted should be exactly like this: P-Asserted-Identity: "xxxxxxx"<sip:YYYYYYYYY77@y.y.y.y:5060>
So actuall there are two problem <tel: have to be changed to <sip: and after the IP the Port :5060 is missing.
My rules actuall looks like this so normally <sip: don`t have to change to <tel: and I also have the port 5060 in the rule:
request ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*)<sip:XXXX77@x.x.x.x>" "P-Asserted-Identity:\1<sip:YYYYYYYY77@y.y.y.y:5060>"
response ANY sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*)<sip:XXXX77@x.x.x.x>" "P-Asserted-Identity:\1<sip:YYYYYYYY77@y.y.y.y:5060>"
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