[cisco-voip] Issue with anonymous calls on a SIP trunk
Andy
andy.carse at gmail.com
Wed Dec 11 13:32:11 EST 2013
Brian,
This is the current config of the Dial-Peers
dial-peer voice 50 voip
description Primary Inbound Dialplan from SIP Trunk to SUB02
preference 1
destination-pattern 44......T
session protocol sipv2
session target ipv4:<SUB02 IP>
voice-class codec 711
voice-class sip outbound-proxy ipv4:<SUB02 IP>
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind media source-interface GigabitEthernet0/2
dtmf-relay rtp-nte
no vad
dial-peer voice 100 voip
description Primary Outbound Dialpeer from CUCM to SIP Trunk
destination-pattern .T
progress_ind setup enable 3
session protocol sipv2
session target sip-server
voice-class codec 711
voice-class sip options-keepalive
dtmf-relay rtp-nte
no vad
Regards Andy
On 10/12/2013 19:46, Brian Meade (brmeade) wrote:
>
> Can you verify if you have your dial-peers set up for media flow-through?
>
> *From:*Andy Carse [mailto:andy.carse at gmail.com]
> *Sent:* Tuesday, December 10, 2013 2:07 PM
> *To:* Brian Meade (brmeade)
> *Cc:* Cisco VoIP List
> *Subject:* RE: [cisco-voip] Issue with anonymous calls on a SIP trunk
>
> Yes it was taken at the providers end of sip trunk.
>
> On 10 Dec 2013 18:34, "Brian Meade (brmeade)" <brmeade at cisco.com
> <mailto:brmeade at cisco.com>> wrote:
>
> 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.
>
> Brian
>
> -----Original Message-----
> From: Andy [mailto:andy.carse at gmail.com <mailto:andy.carse at gmail.com>]
> Sent: Tuesday, December 10, 2013 12:19 PM
> To: Brian Meade (brmeade); Cisco VoIP List
> Subject: Re: [cisco-voip] Issue with anonymous calls on a SIP trunk
>
> Hi Brian,
> I only have a sniffer trace to hand at the moment I've changed the ip
> addressing and the numbers to protect the innocent.
>
> Internet Protocol Version 4, Src: 10.1.1.2 (10.1.1.2), Dst: 10.1.1.1
> (10.1.1.1)
> User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
> Session Initiation Protocol
> Status-Line: SIP/2.0 200 OK
> Status-Code: 200
> [Resent Packet: False]
> Message Header
> Via: SIP/2.0/UDP
> 10.1.1.1:5060;branch=z9hG4bKbjrn5f305g111q46j2j1.1
> Transport: UDP
> Sent-by Address: 10.1.1.1
> Sent-by port: 5060
> Branch: z9hG4bKbjrn5f305g111q46j2j1.1
> From:
> "Anonymous"<sip:anonymous at 10.1.1.1
> <mailto:sip%3Aanonymous at 10.1.1.1>>;tag=140140856-1386321996272-
> SIP Display info: "Anonymous"
> SIP from address: sip:anonymous at 10.1.1.1
> <mailto:sip%3Aanonymous at 10.1.1.1>
> SIP from address User Part: anonymous
> SIP from address Host Part: 10.1.1.1
> SIP tag: 140140856-1386321996272-
> To: "44InboundDDI
> 44InBoundDDI"<sip:44InBoundDDI@"Domain"
> <sip:44InBoundDDI@%22Domain%22>>;tag=585CA458-26EF
> SIP Display info: "44InboundDDI 44InBoundDDI"
> SIP to address: sip:44InBoundDDI@"Domain
> <sip:44InBoundDDI@%22Domain>"
> SIP to address User Part: 44InBoundDDI
> SIP to address Host Part: "Domain"
> SIP tag: 585CA458-26EF
> Date: Fri, 06 Dec 2013 09:26:36 GMT
> Call-ID: BW092636272061213411136895 at 10.81.253.80
> <mailto:BW092636272061213411136895 at 10.81.253.80>
> CSeq: 597521657 INVITE
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE,
> REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
> Allow-Events: telephone-event
> Contact: <sip:44InBoundDDI at 10.1.1.2:5060
> <http://sip:44InBoundDDI@10.1.1.2:5060>>
> Contact-URI: sip:44InBoundDDI at 10.1.1.2:5060
> <http://sip:44InBoundDDI@10.1.1.2:5060>
> Supported: replaces
> Supported: sdp-anat
> Server: Cisco-SIPGateway/IOS-15.3.2.T1
> Supported: timer
> Content-Type: application/sdp
> Content-Disposition: session;handling=required
> Content-Length: 246
> Message Body
> Session Description Protocol
> Session Description Protocol Version (v): 0
> Owner/Creator, Session Id (o): CiscoSystemsSIP-GW-UserAgent
> 9234 6163 IN IP4 10.1.1.2
> Owner Username: CiscoSystemsSIP-GW-UserAgent
> Session ID: 9234
> Session Version: 6163
> Owner Network Type: IN
> Owner Address Type: IP4
> Owner Address: 10.1.1.2
> Session Name (s): SIP Call
> Connection Information (c): IN IP4 "CallManager IP Address"
> Time Description, active time (t): 0 0
> Session Start Time: 0
> Session Stop Time: 0
> Media Description, name and address (m): audio 26000
> RTP/AVP 0 101
> Connection Information (c): IN IP4 "CallManager IP Address"
> Media Attribute (a): rtpmap:0 PCMU/8000
> Media Attribute (a): rtpmap:101 telephone-event/8000
> Media Attribute (a): fmtp:101 0-15
> Media Attribute (a): ptime:20
>
> Regards
>
> Andy
>
> On 10/12/2013 14:36, Brian Meade (brmeade) wrote:
> > Andy,
> >
> > 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.
> >
> > Brian
> >
> > -----Original Message-----
> > From: cisco-voip [mailto:cisco-voip-bounces at puck.nether.net
> <mailto:cisco-voip-bounces at puck.nether.net>] On Behalf
> > Of Andy
> > Sent: Tuesday, December 10, 2013 6:32 AM
> > To: Cisco VoIP List
> > Subject: [cisco-voip] Issue with anonymous calls on a SIP trunk
> >
> > Hi,
> > I have an issue with anonymous (callerid witheld) calls on a SIP
> trunk which I can't figure out.
> >
> > 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.
> > 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.
> >
> > Does anyone have any idea how to fix this?
> >
> > --
> > Regards
> >
> > Andy
> >
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net>
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
>
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