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<div class="moz-cite-prefix">Brian, <br>
This is the current config of the Dial-Peers<br>
<br>
dial-peer voice 50 voip<br>
description Primary Inbound Dialplan from SIP Trunk to SUB02<br>
preference 1<br>
destination-pattern 44......T<br>
session protocol sipv2<br>
session target ipv4:<SUB02 IP><br>
voice-class codec 711<br>
voice-class sip outbound-proxy ipv4:<SUB02 IP><br>
voice-class sip bind control source-interface GigabitEthernet0/2<br>
voice-class sip bind media source-interface GigabitEthernet0/2<br>
dtmf-relay rtp-nte<br>
no vad<br>
<br>
dial-peer voice 100 voip<br>
description Primary Outbound Dialpeer from CUCM to SIP Trunk<br>
destination-pattern .T<br>
progress_ind setup enable 3<br>
session protocol sipv2<br>
session target sip-server<br>
voice-class codec 711<br>
voice-class sip options-keepalive<br>
dtmf-relay rtp-nte<br>
no vad<br>
<br>
Regards
Andy<br>
On 10/12/2013 19:46, Brian Meade (brmeade) wrote:<br>
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<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D">Can
you verify if you have your dial-peers set up for media
flow-through?<o:p></o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span
style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">
Andy Carse [<a class="moz-txt-link-freetext" href="mailto:andy.carse@gmail.com">mailto:andy.carse@gmail.com</a>]
<br>
<b>Sent:</b> Tuesday, December 10, 2013 2:07 PM<br>
<b>To:</b> Brian Meade (brmeade)<br>
<b>Cc:</b> Cisco VoIP List<br>
<b>Subject:</b> RE: [cisco-voip] Issue with anonymous calls
on a SIP trunk<o:p></o:p></span></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p>Yes it was taken at the providers end of sip trunk.<o:p></o:p></p>
<div>
<p class="MsoNormal">On 10 Dec 2013 18:34, "Brian Meade
(brmeade)" <<a moz-do-not-send="true"
href="mailto:brmeade@cisco.com">brmeade@cisco.com</a>>
wrote:<o:p></o:p></p>
<p class="MsoNormal" style="margin-bottom:12.0pt">Was this
capture taken from outside the CUBE? It looks like you
might not be using media flow-through on your dial-peers if
that media IP address isn't getting updated.<br>
<br>
Brian<br>
<br>
-----Original Message-----<br>
From: Andy [mailto:<a moz-do-not-send="true"
href="mailto:andy.carse@gmail.com">andy.carse@gmail.com</a>]<br>
Sent: Tuesday, December 10, 2013 12:19 PM<br>
To: Brian Meade (brmeade); Cisco VoIP List<br>
Subject: Re: [cisco-voip] Issue with anonymous calls on a
SIP trunk<br>
<br>
Hi Brian,<br>
I only have a sniffer trace to hand at the moment I've
changed the ip addressing and the numbers to protect the
innocent.<br>
<br>
Internet Protocol Version 4, Src: 10.1.1.2 (10.1.1.2), Dst:
10.1.1.1<br>
(10.1.1.1)<br>
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip
(5060) Session Initiation Protocol<br>
Status-Line: SIP/2.0 200 OK<br>
Status-Code: 200<br>
[Resent Packet: False]<br>
Message Header<br>
Via: SIP/2.0/UDP
10.1.1.1:5060;branch=z9hG4bKbjrn5f305g111q46j2j1.1<br>
Transport: UDP<br>
Sent-by Address: 10.1.1.1<br>
Sent-by port: 5060<br>
Branch: z9hG4bKbjrn5f305g111q46j2j1.1<br>
From:<br>
"Anonymous"<<a moz-do-not-send="true"
href="mailto:sip%3Aanonymous@10.1.1.1">sip:anonymous@10.1.1.1</a>>;tag=140140856-1386321996272-<br>
SIP Display info: "Anonymous"<br>
SIP from address: <a moz-do-not-send="true"
href="mailto:sip%3Aanonymous@10.1.1.1">sip:anonymous@10.1.1.1</a><br>
SIP from address User Part: anonymous<br>
SIP from address Host Part: 10.1.1.1<br>
SIP tag: 140140856-1386321996272-<br>
To: "44InboundDDI<br>
44InBoundDDI"<<a moz-do-not-send="true"
href="sip:44InBoundDDI@%22Domain%22">sip:44InBoundDDI@"Domain"</a>>;tag=585CA458-26EF<br>
SIP Display info: "44InboundDDI 44InBoundDDI"<br>
SIP to address: <a moz-do-not-send="true"
href="sip:44InBoundDDI@%22Domain">sip:44InBoundDDI@"Domain</a>"<br>
SIP to address User Part: 44InBoundDDI<br>
SIP to address Host Part: "Domain"<br>
SIP tag: 585CA458-26EF<br>
Date: Fri, 06 Dec 2013 09:26:36 GMT<br>
Call-ID: <a moz-do-not-send="true"
href="mailto:BW092636272061213411136895@10.81.253.80">BW092636272061213411136895@10.81.253.80</a><br>
CSeq: 597521657 INVITE<br>
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK,
UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER<br>
Allow-Events: telephone-event<br>
Contact: <<a moz-do-not-send="true"
href="http://sip:44InBoundDDI@10.1.1.2:5060"
target="_blank">sip:44InBoundDDI@10.1.1.2:5060</a>><br>
Contact-URI: <a moz-do-not-send="true"
href="http://sip:44InBoundDDI@10.1.1.2:5060"
target="_blank">
sip:44InBoundDDI@10.1.1.2:5060</a><br>
Supported: replaces<br>
Supported: sdp-anat<br>
Server: Cisco-SIPGateway/IOS-15.3.2.T1<br>
Supported: timer<br>
Content-Type: application/sdp<br>
Content-Disposition: session;handling=required<br>
Content-Length: 246<br>
Message Body<br>
Session Description Protocol<br>
Session Description Protocol Version (v): 0<br>
Owner/Creator, Session Id (o):
CiscoSystemsSIP-GW-UserAgent<br>
9234 6163 IN IP4 10.1.1.2<br>
Owner Username:
CiscoSystemsSIP-GW-UserAgent<br>
Session ID: 9234<br>
Session Version: 6163<br>
Owner Network Type: IN<br>
Owner Address Type: IP4<br>
Owner Address: 10.1.1.2<br>
Session Name (s): SIP Call<br>
Connection Information (c): IN IP4 "CallManager
IP Address"<br>
Time Description, active time (t): 0 0<br>
Session Start Time: 0<br>
Session Stop Time: 0<br>
Media Description, name and address (m): audio
26000 RTP/AVP 0 101<br>
Connection Information (c): IN IP4 "CallManager
IP Address"<br>
Media Attribute (a): rtpmap:0 PCMU/8000<br>
Media Attribute (a): rtpmap:101
telephone-event/8000<br>
Media Attribute (a): fmtp:101 0-15<br>
Media Attribute (a): ptime:20<br>
<br>
Regards<br>
<br>
Andy<br>
<br>
On 10/12/2013 14:36, Brian Meade (brmeade) wrote:<br>
> Andy,<br>
><br>
> Can you copy what the Update message looks like so we
can see what header the CUCM IP address is in? You should
be able to use a SIP Profile on the CUBE to change this to
the CUBE's external IP address.<br>
><br>
> Brian<br>
><br>
> -----Original Message-----<br>
> From: cisco-voip [mailto:<a moz-do-not-send="true"
href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>]
On Behalf<br>
> Of Andy<br>
> Sent: Tuesday, December 10, 2013 6:32 AM<br>
> To: Cisco VoIP List<br>
> Subject: [cisco-voip] Issue with anonymous calls on a
SIP trunk<br>
><br>
> Hi,<br>
> I have an issue with anonymous (callerid witheld) calls
on a SIP trunk which I can't figure out.<br>
><br>
> Call comes in over sip trunk via a cube to cucm, if the
callerid is know then the call gets placed to the dialed
number ok.<br>
> But if the number is withheld on the inbound call leg
their is an additional update message which contains the
CUCM ip address, but the SIP provider is unable to route to
this address so one way voice is the result.<br>
><br>
> Does anyone have any idea how to fix this?<br>
><br>
> --<br>
> Regards<br>
><br>
> Andy<br>
><br>
> _______________________________________________<br>
> cisco-voip mailing list<br>
> <a moz-do-not-send="true"
href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
> <a moz-do-not-send="true"
href="https://puck.nether.net/mailman/listinfo/cisco-voip"
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><o:p></o:p></p>
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