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

Sat Mar 22 09:49:06 EDT 2014

If it's an higher ios,  let's first allow it to accept traffic by,
disabling toll fraud, or adding a trust list.

Conf t
Voice service voip
No ip address trust list authenticate

In addition to this any voip dialpeer with, incoming called-number & a list
of codecs, would help.

Last but not least sip control & media binding is needed if router got more
than one ip interfaces.
 On 22-Mar-2014 2:10 am, "Jason Aarons (AM)" <jason.aarons at dimensiondata.com>
wrote:

> 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>
>
> 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 <msaskin at gmail.com>
>
> *Sent:* Friday, March 21, 2014 3:20 PM
>
> *To:* Jason Aarons (AM) <jason.aarons at dimensiondata.com>
>
> *Cc:* Reto Gassmann <voip at mrga.ch> ; 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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>
>
>
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