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

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.

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

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.