11-06-2007 11:36 AM
I need to work with interoperability between Cisco AS5300XM and Level 3 over VoIP. But need ideas about the Cisco equipment setup. Level 3 said "you need E.164 numbering plan and check the Call-ID" .
About the E.164 insert the symbol "+" plus the 11 digit 1+area code+number
Please let me know any ideas
thanks
11-06-2007 12:01 PM
Hi,
you can use a translation-rule like:
voice translation-profile e164
translate called 100
voice translation-rule 100
rule 1 /^1/ /+1/
dial-peer voice 33 voip
destination-pattern 1[2-9][0-8][0-9][2-9]......
translation-profile outgoing e164
That will add the required + for US domestic calls. Repeat in other DPs as necessary for other destinations.
You may even try without the +, just use 011 for example. Chances are it will work anyway :)
Hope this helps, please rate post if it does!
11-06-2007 12:56 PM
Hi
thanks for your help
I used the following setup and don't work
For Incoming
dial-peer voice 222 pots
destination-pattern +T
port 1/0:D
prefix 1
For Outbound
voice translation-profile e164
translate called 1
translation-rule 1
Rule 1 ^1 +1
dial-peer voice 203 voip
translation-profile outgoing e164
destination-pattern 1.......... (11 digit)
session protocol sipv2
session target ipv4:x.x.x.x:5060 (IP Level3)
session transport udp
dtmf-relay rtp-nte
clid network-number +xxxxxxxxxx (my number)
My equipment is
Cisco AS5350XM
Version 12.4(11)T
thanks
Erick
11-06-2007 01:57 PM
Hi,
please configure "voice translation-rule 1" and not "translation-rule 1" as these are different things.
Then when checking, please enable "debug ccsip message" and "term mon", in case of problems you can post the output here.
Note the reason I made an example with a more complicated destination-pattern than just 1 followed by 10 dots, it's because it enforces the NANP and prevents user misdialing. That is always a good thing in telephony.
Also note your DP 222 for inbound seems too relaxed in accepting everything and just sending out.
11-06-2007 04:44 PM
Hi,
We tried several calls to Level 3 equipment , the signaling passed well but not the audio
Another thing if you can help me with understanding this call-id , this is the string that they are requesting us to send to generate call to us ( Level 3 to our equipment)
Example Call-Ids:
DEN05020041014230534123123@209.244.48.214
56393400ad0a02108000000e0c6eab9a@64.24.128.4
I get this for the Cisco box (AS5350XM)
Nov 6 18:31:52.550: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:+17865221044@209.247.16.1:5060 SIP/2.0
Via: SIP/2.0/UDP 190.7.193.85:5060;branch=z9hG4bK1B04BD
Remote-Party-ID: <>;party=calling;screen=yes;privacy=off>
From: <>;tag=F3098510-1337>
To: <>>
Date: Tue, 06 Nov 2007 18:31:52 GMT
Call-ID: 60F519DC-8BCD11DC-89D680BC-B3FDBCC9@190.7.193.85
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1626595571-2345472476-2173108246-2641290602
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1194373912
Contact: <>>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 294
v=0
o=CiscoSystemsSIP-GW-UserAgent 3437 4571 IN IP4 190.7.193.85
s=SIP Call
c=IN IP4 190.7.193.85
t=0 0
m=audio 18244 RTP/AVP 18 101 19
c=IN IP4 190.7.193.85
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20
Nov 6 18:31:52.666: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 190.7.193.85:5060;branch=z9hG4bK1B04BD
From: <>;tag=F3098510-1337>
To: <>>
Call-ID: 60F519DC-8BCD11DC-89D680BC-B3FDBCC9@190.7.193.85
CSeq: 101 INVITE
Content-Length: 0
Nov 6 18:31:55.462: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 190.7.193.85:5060;branch=z9hG4bK1B04BD
From: <>;tag=F3098510-1337>
To: <>;tag=VPST50603522629633>
Call-ID: 60F519DC-8BCD11DC-89D680BC-B3FDBCC9@190.7.193.85
CSeq: 101 INVITE
Contact: <>>
MIME-Version: 1.0
Content-Type: multipart/mixed;boundary=level3-viper-boundary
Content-Length: 392
--level3-viper-boundary
Content-Type: application/sdp
v=0
o=- 1194395489 1194395490 IN IP4 209.247.5.45
s=-
c=IN IP4 209.247.5.45
t=0 0
m=audio 60238 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
--level3-viper-boundary
Content-Type: application/isup;version=ansi
Content-Disposition: session;handling=optional
Best Regards
11-07-2007 08:28 AM
Hi,
do you have any FW or NAT between the AS5300 and the internet ?
I can't quite understand the call-id issue, normall the call-id is dynamically generated, for call generated toward you what matters is the called number more than anything else.
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