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

Fri Mar 21 16:38:25 EDT 2014

So this was a outbound dialer call that dialed sip:10.1.1.1

I don’t see where it dialed a number.

You can test incoming called-number . on a POTS.

(if 10.1.1.1 was the gateway with a PRI)

There are some examples here as well as helping to understand how dial peers work.   If you forget about UCCE and just try to call out from CallManager using SIP does it work…
http://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/14074-in-dial-peer-match.html

From: Reto Gassmann [mailto:voip at mrga.ch]
Sent: Friday, March 21, 2014 4:22 PM
To: Matthew Saskin; Jason Aarons (AM)
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Gateway configuration for UCCCE SIP Outbound Dialer

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<sip:sipping@[PG%20IP]:58800>>
To: <sip:[GW IP]<sip:[GW%20IP]>>
From: <sip:sipping@[PG IP]<sip:sipping@[PG%20IP]>>;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]<sip:sipping@[PG%20IP]>>;tag=ea21336d
To: <sip:[GW IP]<sip:[GW%20IP]>>;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<mailto:msaskin at gmail.com>
Sent: Friday, March 21, 2014 3:20 PM
To: Jason Aarons (AM)<mailto:jason.aarons at dimensiondata.com>
Cc: Reto Gassmann<mailto:voip at mrga.ch> ; cisco-voip at puck.nether.net<mailto: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<mailto:msaskin at gmail.com>
203-253-9571

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