[cisco-voip] Gateway configuration for UCCCE SIP Outbound Dialer

[Reto Gassmann]

Hi Matthew, Hi Jason

I set up a ccsip debug and got the follwing result:

Mar 21 09:39:37 MEZ: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:[GW IP] SIP/2.0
Via: SIP/2.0/UDP [PG IP]:58800;branch=z9hG4bK-d8754z-4715fa57f237d759-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:sipping@[PG IP]:58800>
To: <sip:[GW IP]>
From: <sip:sipping@[PG IP]>;tag=ea21336d
Call-ID: b5770f02-1a337504-723c5606-9545a22b
CSeq: 1 OPTIONS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, NOTIFY, PRACK, REFER, NOTIFY, OPTIONS
Supported: timer, resource-priority, replaces
Content-Length: 0


Mar 21 09:39:37 MEZ: //58166/29ABFC7EB86B/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP [PG IP]:58800;branch=z9hG4bK-d8754z-4715fa57f237d759-1---d8754z-;rport
From: <sip:sipping@[PG IP]>;tag=ea21336d
To: <sip:[GW IP]>;tag=12ABD6AC-11F5
Date: Fri, 21 Mar 2014 08:39:37 GMT
Call-ID: b5770f02-1a337504-723c5606-9545a22b
Server: Cisco-SIPGateway/IOS-15.2.3.T4
CSeq: 1 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 370

v=0
o=CiscoSystemsSIP-GW-UserAgent 693 5538 IN IP4 [GW IP]
s=SIP Call
c=IN IP4 [GW IP]
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 [Gw IP]
m=image 0 udptl t38
c=IN IP4 [GW IP]
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy


Mar 21 09:32:07 MEZ: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [PG IP]:58800, local_address:[ - ]
Mar 21 09:32:07 MEZ: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
Mar 21 09:32:07 MEZ: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
Mar 21 09:32:07 MEZ: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x32174578) with key=[49894] to table
Mar 21 09:32:07 MEZ: //-1/000000000000/SIP/Info/ccsip_offer_ans_init:
Mar 21 09:32:07 MEZ: //-1/000000000000/SIP/Info/ccsip_iwf_init:
Mar 21 09:32:07 MEZ: //-1/000000000000/SIP/Info/ccsip_ipip_media_service_init:
Mar 21 09:32:07 MEZ: //-1/000000000000/SIP/Info/sipSPI_ipip_vcc_Initialization:  Entry...
Mar 21 09:32:07 MEZ: //-1/000000000000/SIP/Info/ccsip_ipip_media_forking_init: MF: Queue is initialised..
Mar 21 09:32:07 MEZ: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
Mar 21 09:32:07 MEZ: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: ip_best_local_address [GW IP] for SIP
Mar 21 09:32:07 MEZ: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr [GW IP]
Mar 21 09:32:07 MEZ: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone MEZ to SIP default timezone = GMT
Mar 21 09:32:07 MEZ: //58064/1F65EE49B7FC/SIP/Info/ccsipInitPldCallingInfo: non-numeric calling number: sipping
Mar 21 09:32:07 MEZ: //58064/1F65EE49B7FC/SIP/Info/sipSPIGetViaHostInURLFormat: VIA URL Conversion failed for
Mar 21 09:32:07 MEZ: //58064/1F65EE49B7FC/SIP/Info/sipSPIGetShrlPeer: Try match incoming dialpeer for Calling number: : sipping
Mar 21 09:32:07 MEZ: //58064/1F65EE49B7FC/SIP/Info/sipSPIGetCallConfig: No match found for P-Called-Party-ID
Mar 21 09:32:07 MEZ: //58064/1F65EE49B7FC/SIP/Info/sipSPIGetCallConfig: Precondition tag absent in Require/Supported header
Mar 21 09:32:07 MEZ: //58064/1F65EE49B7FC/SIP/Info/sipSPIGetCallConfig: Media Antitrombone disabled
Mar 21 09:32:07 MEZ: //58064/1F65EE49B7FC/SIP/Info/sipSPISetMediaFlowMode: Storing the configured mode as FLOW-THROUGH
Mar 21 09:32:07 MEZ: //58064/1F65EE49B7FC/SIP/Info/sipSPISetMediaFlowMode: xcoder high-density disabled
Mar 21 09:32:07 MEZ: //58064/1F65EE49B7FC/SIP/Info/sipSPISetMediaFlowMode: Flow Mode set to FLOW_THROUGH
Mar 21 09:32:07 MEZ: //58064/1F65EE49B7FC/SIP/Info/sipSPIGetCallConfig: Non dial peer leg - using RTP Supported Codecs
Mar 21 09:32:07 MEZ: //58064/1F65EE49B7FC/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 18
Mar 21 09:32:07 MEZ: //58064/1F65EE49B7FC/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 0
Mar 21 09:32:07 MEZ: //58064/1F65EE49B7FC/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 8
Mar 21 09:32:07 MEZ: //58064/1F65EE49B7FC/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 9
Mar 21 09:32:07 MEZ: //58064/1F65EE49B7FC/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 4
Mar 21 09:32:07 MEZ: //58064/1F65EE49B7FC/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 2
Mar 21 09:32:07 MEZ: //58064/1F65EE49B7FC/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 15
Mar 21 09:32:07 MEZ: //58064/1F65EE49B7FC/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 255


It seems that the there is a inbound dialpeer missing. Do you know how to set up a inbound dialüpeer if i have a calling like “sipping”?

Thanks for help

Regards Reto


From: Matthew Saskin 
Sent: Friday, March 21, 2014 3:20 PM
To: Jason Aarons (AM) 
Cc: Reto Gassmann ; cisco-voip at puck.nether.net 
Subject: Re: [cisco-voip] Gateway configuration for UCCCE SIP Outbound Dialer

Do you have an inbound SIP dial-peer to accept the calls from the PG?  If not, call may be hitting a default dial peer and behaving badly.  I agree with Jason's suggestion - start with some gateway level ccsip and dial-peer debugs and see what's happening.


Matthew Saskin
msaskin at gmail.com
203-253-9571




On Thu, Mar 20, 2014 at 5:25 PM, Jason Aarons (AM) <jason.aarons at dimensiondata.com> wrote:

  Did you debug ccsip messages (SIP) and then correlate, debug voip dial peer (h323), debug voip ccapi inout, etc



  http://www.cisco.com/c/en/us/support/docs/voice/h323/14081-voip-debugcalls.html





  From: cisco-voip [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Reto Gassmann
  Sent: Thursday, March 20, 2014 5:09 PM
  To: cisco-voip at puck.nether.net
  Subject: [cisco-voip] Gateway configuration for UCCCE SIP Outbound Dialer





  Hello Group



  we try to set up the SIP Outbounddialer with a UCCE 9.

  The UCCE part is working and the PG is sending SIP to the Gateway. However, the Gateway (2921 IOS 15.2(3) T4) does not accept the SIP requests.

  The gateway is also used for PSTN calls to the CUCM. We habe dialpeers set up to the PRI (PSTN) as well as to the CUCM (h.323).



  dial-peer voice 9900 voip

  tone ringback alert-no-PI

  description  to CUCM PreProd

  preference 2

  destination-pattern 99[0-7].

  progress_ind setup enable 3

  session target ipv4: [CUCM IP]

  voice-class codec 3

  voice-class h323 1

  dtmf-relay h245-alphanumeric

  playout-delay nominal 50

  no vad



  Has anyone a sample SIP configuartion of a Gateway that works with the SIP Outbound Dialer?

  we have added this config to the gateway



  voice service voip

  ip address trusted list

    ipv4 [UCCE PG IP]

  allow-connections h323 to h323

  allow-connections h323 to sip

  allow-connections sip to h323

  allow-connections sip to sip

  sip

    rel1xx supported 100rel



  Thanks for you help



  Reto



  itevomcid 


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