Voice gateway, SIP Options ping without SDP

Hi all.
I see following SIP options call-flow for sip options ping:
17907581: Jan 16 08:11:31.057: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:10.250.20.25:5060 SIP/2.0
Via: SIP/2.0/UDP 10.11.3.17:9256;branch=z9hG4bKabuf3p5e3es57i595u9viaf77
Call-ID: [email protected]
From: <sip:[email protected]>;tag=sbc0800uf7upc43
To: <sip:10.250.20.25>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0
17907583: Jan 16 08:11:31.057: //28111388/1CCF073F9ED3/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.3.17:9256;branch=z9hG4bKabuf3p5e3es57i595u9viaf77
From: <sip:[email protected]>;tag=sbc0800uf7upc43
To: <sip:10.250.20.25>;tag=BC08FBDC-1F16
Date: Fri, 16 Jan 2015 08:11:31 GMT
Call-ID: [email protected]
Server: Cisco-SIPGateway/IOS-15.4.20141104.060737.
CSeq: 1 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 363
v=0
o=CiscoSystemsSIP-GW-UserAgent 1616 9170 IN IP4 10.13.4.43
s=SIP Call
c=IN IP4 10.13.4.43
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 10.13.4.43
m=image 0 udptl t38
c=IN IP4 10.13.4.43
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy
I know that this is normal behavior for voice gateway(3925 in our case).
Is it possible to disable SDP in 200 OK message going out to ITSP. They said that their PBX can't recognize 200 ok with SDP as reply to sip Options message.

Oleh,
I am not sure you can do this..
RFC3261 states the following:
The SIP method OPTIONS allows a UA to query another UA or a proxy
   server as to its capabilities. This allows a client to discover
   information about the supported methods, content types, extensions,
   codecs, etc. without "ringing" the other party. For example, before
   a client inserts a Require header field into an INVITE listing an
   option that it is not certain the destination UAS supports, the
   client can query the destination UAS with an OPTIONS to see if this
   option is returned in a Supported header field. All UAs MUST support
   the OPTIONS method.
If the response to an OPTIONS is generated by a proxy server, the
   proxy returns a 200 (OK), listing the capabilities of the server.
   The response does not contain a message body.
So it looks like your ITSP hasn't designed their system based on this RFC. When you send an OPTIONs message, the response has to include the capabilities the other party can support..

Similar Messages

Best way to implement SIP Options Pings on a SIP Trunk

I wanted to see if anyone had suggestions on the best way to configure SIP Options Pings.
Typically I would configure them per dial-peer. However, I really want to do it per destination IP address. I do not want a SIP Options ping for every single dial-peer being sent out every X seconds.
Example:
In my case I have 4 SIP trunks in the same CUBE. Each pointing to a different destination IP. There are 6 dial-peers per SIP trunk. I really do not want 24 option pings going out every X seconds. I guess I've never actually did a debug to see how many pings are going out at a time but I am assuming it sends one for each dial-peer or does it?
If I am correct in my assumption, is there way to only send one ping per destination IP and if that single IP goes unresponsive then all 6 dial-peers go down?

Hello,
The OOD option ping is sent per dial-peer to the destination.
Restrictions
•The Cisco Unified Border Element OOD Options ping feature can only be configured at the VoIP Dial-peer level.
•All dial peers start in an active (not busied out) state on a router boot or reboot.
•If a dial-peer has both an outbound proxy and a session target configured, the OOD options ping is sent to the outbound proxy address first.
•Though multiple dial-peers may point to the same SIP server IP address, an independent OOD options ping is sent for each dial-peer.
•If a SIP server is configured as a DNS hostname, OOD Options pings are sent to all the returned addresses until a response is received.
•Configuration for Cisco Unified Border Element OOD and TDM Gateway OOD are different, but can co-exist.
//Suresh
Please rate all the useful posts.

SIP Options Ping

Cisco CME to Cisco UBE Options ping no-worky....
Did anyone get SIP options pings working on a pure Cisco router (UBE to UBE and UBE to CME) environment? I have a pure Cisco 15.1 network that we are playing with option pings on. I see pings go out and 200 OK responses, but dial-peers are still busying out. Have tried both default and example settings from Cisco UBE deployment guides. Routers are working with options pings from external ITSPs, so know options replies are working and are supported. So, what's the magic sauce on Cisco IOS 15.1 for SIP options-keepalive?
dial-peer voice 40221 voip
corlist outgoing peer-internal
description \\\ 4385 - 4389 \\\
translation-profile outgoing ToCUCME_NANP_PrefixedE164
preference 1
destination-pattern 438[5-9]
session protocol sipv2
session target ipv4:10.5.5.1
voice-class sip early-offer forced
voice-class sip options-keepalive up-interval 12 down-interval 65 retry 3
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-throug
no vad

There are no firewalls or NAT between and when using "debug ccsip message" we see option sent and received and replies on both sides.

SIP Options PING (CVP SIP Server Groups - Heartbeat)

The CVP Operations Console has a feature to enable SIP ping Options.
If you configure a SIP server group and enable heartbeats, the CVP Call Server sends a SIP ping option ever X seconds.
We have a SIP Server group for the CUCM servers and a group for the VXML Gateways.
The Cisco CUCM replies with a SIP 200 OK to these SIP 'ping' Options.
However the VXML Gateway does not. Does anyone know why not and if its possible to enable a gateway to reply to these SIP options?
Below is documentation on how to configure the Cisco Gateway to SEND ping options, but we don't that. We just need it to reply to the Ping OPTIONS sent by CVP.
For calls that originate from CUCM to CVP, they need to use the VXML Gateway SIP server group, (as it cannot route to the originator for VXML treatment).
http://www.cisco.com/c/en/us/td/docs/ios/voice/cube/configuration/guide/vb_book/vb_book/vb_9321.html
See screen shot where you can see replies from the CUCM servers (.200, .201) but NOT from VXML Gateway (.250).
Regards,
Gerry

Hello Gerry,
You may need to enable the SIP Traces in the voice Gateway and have a look how exactly gateway is processing and why 200 OK is not sent.
Can you share the IOS Version on the Gateway ?
Look at the below link and list of Resolved Caveats in 15.2(4)M, one of the caveat looks like your scenario. But i would recommend first look into Sip traces.
http://www.cisco.com/c/en/us/td/docs/ios/15_2m_and_t/release/notes/15_2m_and_t/152-4MCAVS.html
CSCtx79318
Symptoms: OGW fails to send 200 OK response for OPTION.
Conditions: The symptom is observed with 200 OK response for OPTION in Cisco IOS interim Release 15.2(02.16)T.
Workaround: There is no workaround.
Regards,
Senthil

Cisco 2911 Voice Gateway SIP PSTN Calls Fail

Hello All,
    I am having trouble with outboud SIP PSTN calls through a Cisco 2911 Voice Gateway. 2911 VG terminates PSTN SIP Traffic and connects to Avaya CS1000M via QSIG PRI Trunks. When calls are attempted outbound fron the PBX the caller gets a fast busy. Debug ISDN q931 shows the call hitting the 2911 properly, debug voip ccapi inout shows the call matching the correct dial peers and debug ccsip shows the invite to the PSTN Provider SBC, however within the invite the "from" address incorrectly shows the calling number with the provider SBC address (see below). does anyone have any insight on how to correct this? Attached are VG config and Debug isdn q931, voip ccapi inout, ccsip messages and ccsip call. Thanks in advance for any help!!
From: <sip:[email protected]>:tag=6166CDC4-882
To: <sip:[email protected]>
Shawn C. Smith

i have same problem my cucm ip is 192.168.200.53
my Voice Gateway is SIP by ip 192.168.200.86 for internal
and 172.29.7.94
and my SIP Server is 10.208.9.69
if its oky can yuo take a look at my problem please
this is the syslog from debug
May 30 20:19:34.284: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: <sip:192.168.200.53:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
Session-Expires: 1800
P-Asserted-Identity: "Aysar Mohamed" <sip:[email protected]>
Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0
May 30 20:19:34.284: //-1/B5A494800000/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=2217156
   ----- ccCallInfo IE subfields -----
   cisco-ani=2217156
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=90555769123
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
May 30 20:19:34.288: //-1/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x30CF41D4, Call Info(
   Calling Number=2217156,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=90555769123(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=0, Progress Indication=NULL(0), Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=465
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :cc_get_feature_vsa malloc success
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: cc_get_feature_vsa count is 1
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832953048,feature_id:85
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=90555769123(TON=Unknown, NPI=Unknown))
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
   Event=0x2B82D890
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
   Try with the demoted called number 90555769123
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetContext:
   Context=0x2ABC2E44
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 465 with tag 0 to app "_ManagedAppProcess_Default"
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=TRUE, Mode=0,
   Outgoing Dial-peer=802, Params=0x2ABC19D4, Progress Indication=NULL(0)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
   In: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCheckClipClir:
   Out: Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
   Destination Pattern=9T, Called Number=0555769123, Digit Strip=FALSE
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccCallSetupRequest:
   Calling Number=2217156(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=0555769123(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=Aysar Mohamed
   Account Number=2217156, Final Destination Flag=TRUE,
   Guid=B5A49480-0001-0000-0000-11C535C8A8C0, Outgoing Dial-peer=802
May 30 20:19:34.288: //465/B5A494800000/CCAPI/cc_api_display_ie_subfields:
   ccCallSetupRequest:
   cisco-username=2217156
   ----- ccCallInfo IE subfields -----
   cisco-ani=2217156
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=0555769123
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
May 30 20:19:34.288: //465/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
   Interface=0x30CF41D4, Interface Type=3, Destination=, Mode=0x0,
   Call Params(Calling Number=2217156,(Calling Name=Aysar Mohamed)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=0555769123(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
   Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=802, Call Count On=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :cc_get_feature_vsa malloc success
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: cc_get_feature_vsa count is 2
May 30 20:19:34.288: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
May 30 20:19:34.288: :FEATURE_VSA attributes are: feature_name:0,feature_time:832952824,feature_id:86
May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccIFCallSetupRequestPrivate:
   SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
May 30 20:19:34.292: //466/B5A494800000/CCAPI/ccCallSetContext:
   Context=0x2ABC1984
May 30 20:19:34.292: //465/B5A494800000/CCAPI/ccSaveDialpeerTag:
   Outgoing Dial-peer=802
May 30 20:19:34.292: //466/B5A494800000/CCAPI/cc_api_call_proceeding:
   Interface=0x30CF41D4, Progress Indication=NULL(0)
May 30 20:19:34.292: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
May 30 20:19:34.292: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Remote-Party-ID: "Aysar Mohamed" <sip:[email protected]>;party=calling;screen=yes;privacy=off
From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3047462016-0000065536-0000004549-0902342848
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1401481174
Contact: <sip:[email protected]:5060>
Call-Info: <sip:172.29.7.94:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: kpml, telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Length: 0
May 30 20:19:34.300: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>
CSeq: 101 INVITE
Content-Length: 0
May 30 20:19:34.612: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
CSeq: 101 INVITE
Contact: <sip:[email protected]:5060;user=phone>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Content-Length: 328
Content-Type: application/sdp
v=0
o=- 17192647 17192647 IN IP4 10.208.9.69
s=SBC call
c=IN IP4 10.208.9.69
t=0 0
m=audio 39910 RTP/AVP 8 0 102 102 18 116
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 AMR/8000
a=rtpmap:102 AMR/8000
a=rtpmap:18 G729/8000
a=rtpmap:116 telephone-event/8000
a=ptime:5
a=fmtp:116 0-15
a=fmtp:18 annexb=yes
May 30 20:19:34.612: %SIP-3-UNSUPPORTED: Unsupported ptime value
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
   Destination Interface=0x0, Destination Call Id=-1, Source Call Id=466,
   Caps(Codec=0x2, Fax Rate=0x2, Vad=0x1,
   Modem=0x0, Codec Bytes=160, Signal Type=2)
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_caps_ind:
   Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
   Playout Max=1000(ms), Fax Nom=300(ms))
May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
   Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
   Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
   Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/cc_api_caps_ack:
   Destination Interface=0x0, Destination Call Id=-1, Source Call Id=465,
   Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),
   Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=3882)
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event=170, Call Id=466
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event Is Sent To Conferenced SPI(s) Directly
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event=98, Call Id=466
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_event_indication:
   Event Is Sent To Conferenced SPI(s) Directly
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
   Interface=0x30CF41D4, Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1),
   Cause Value=0
May 30 20:19:34.612: //466/B5A494800000/CCAPI/cc_api_call_cut_progress:
   Call Entry(Responsed=TRUE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
   Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1), Cause Value=0
   Voice Call Send Alert=FALSE, Call Entry(Alert Sent=FALSE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccCallCutProgress:
   Call Entry(Responsed=TRUE)
May 30 20:19:34.612: //465/B5A494800000/CCAPI/ccConferenceCreate:
   (confID=0x30C11410, callID1=0x1D1, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
   (confID=0x30C11410, callID2=0x1D2, gcid=8C9E3127-E76E11E3-8274BE8C-EC3B12A0, tag=0x0)
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
   Conference Id=0x30C11410, Call Id1=465, Call Id2=466, Tag=0x0
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
May 30 20:19:34.616: ccConferenceCreate: ret1=0, codecMask1=2, bytes1=160, negot1=0, dtmf1=0
                    ret2=0, codecMask2=2, bytes2=160, negot2=1, dtmf2=6,
                    tx_dynamic_pt1=0, rx_dynamic_pt1=0, codec_mode1=0, params_bitmap1 =0
                    tx_dynamic_pt2=8, rx_dynamic_pt2=8, codec_mode2=0, params_bitmap2 =0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
   delay media to slow start case, codec negotation is not done
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_api_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
   Destination Call Id=466, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
May 30 20:19:34.616: cc_api_get_xcode_stream : 4702
May 30 20:19:34.616: //466/B5A494800000/CCAPI/cc_api_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.616: //465/B5A494800000/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x16, Destination Call Id=466)
May 30 20:19:34.616: //466/B5A494800000/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x16, Destination Call Id=465)
May 30 20:19:34.616: //465/B5A494800000/CCAPI/cc_process_notify_bridge_done:
   Conference Id=0x16, Call Id1=465, Call Id2=466
May 30 20:19:34.616: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:[email protected]>;party=called;screen=yes;privacy=off
Contact: <sip:[email protected]:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 233
v=0
o=CiscoSystemsSIP-GW-UserAgent 2639 5276 IN IP4 192.168.200.86
s=SIP Call
c=IN IP4 192.168.200.86
t=0 0
m=audio 18288 RTP/AVP 8 0 19
c=IN IP4 192.168.200.86
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:19 CN/8000
May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
Record-Route: <sip:10.208.9.69:5060;transport=udp;lr>
Call-ID: [email protected]
From: "Aysar Mohamed"<sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
CSeq: 101 INVITE
Reason: Q.850;cause=127;text="interworking unspecified"
Warning: 399 - "SoftX3000 R601-CCU Rel POS:[3103] Release from CR"
Content-Length: 0
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
   Cause Value=41, Interface=0x30CF41D4, Call Id=466
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=41, Retry Count=0)
May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccCallReleaseResources:
   release reserved xcoding resource.
May 30 20:19:34.680: //466/B5A494800000/CCAPI/ccCallSetAAA_Accounting:
   Accounting=0, Call Id=466
May 30 20:19:34.680: //465/B5A494800000/CCAPI/ccConferenceDestroy:
   Conference Id=0x16, Tag=0x0
May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=465,
   Destination Call Id=466, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //466/B5A494800000/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //465/B5A494800000/CCAPI/cc_generic_bridge_done:
   Conference Id=0x16, Source Interface=0x30CF41D4, Source Call Id=466,
   Destination Call Id=465, Disposition=0x0, Tag=0x0
May 30 20:19:34.680: //466/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.7.94:5060;branch=z9hG4bK461C
From: "Aysar Mohamed" <sip:[email protected]>;tag=7394E4-1898
To: <sip:[email protected]>;tag=sbc0806eppk5yip-CC-57
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: kpml, telephone-event
Content-Length: 0
May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=41)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=466,
   Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
May 30 20:19:34.684: //466/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.684: :cc_free_feature_vsa freeing 31A5D9F0
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.684: vsacount in free is 1
May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
May 30 20:19:34.684: //465/B5A494800000/CCAPI/ccCallDisconnect:
   Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)
May 30 20:19:34.684: //465/B5A494800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=41
Content-Length: 0
May 30 20:19:34.684: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.53:5060;branch=z9hG4bK3bd451bf17e0
From: "Aysar Mohamed" <sip:[email protected]>;tag=37693~244641b0-36ac-434c-91c1-823f25a68b28-18299026
To: <sip:[email protected]>;tag=739628-1BDB
Date: Fri, 30 May 2014 20:19:34 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0
May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x30CF41D4, Tag=0x0, Call Id=465,
   Call Entry(Disconnect Cause=41, Voice Class Cause Code=0, Retry Count=0)
May 30 20:19:34.688: //465/B5A494800000/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.688: :cc_free_feature_vsa freeing 31A5DAD0
May 30 20:19:34.688: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
May 30 20:19:34.688: vsacount in free is 0
May 30 20:19:36.044: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.29.7.94:5060 SIP/2.0
Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
To: <sip:172.29.7.94>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0
May 30 20:19:36.048: //467/8DAABF6C8278/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.208.9.69:5060;branch=z9hG4bKmisco3ykfiooegpygsphkocp1T20326
From: <sip:172.29.7.94:5060>;tag=sbc0803k1pyk51o
To: <sip:172.29.7.94>;tag=739BBC-1CE2
Date: Fri, 30 May 2014 20:19:36 GMT
Call-ID: isbcfemyk1p1mkteets1tcmi53eeehfhikcp@SoftX3000
Server: Cisco-SIPGateway/IOS-12.x
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: 446
v=0
o=CiscoSystemsSIP-GW-UserAgent 3496 1601 IN IP4 172.29.7.94
s=SIP Call
c=IN IP4 172.29.7.94
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 172.29.7.94
m=image 0 udptl t38
c=IN IP4 172.29.7.94
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy
My SIP GW internal ip address is 192.168.200.86
and the Public IP is : 172.29.7.94
My CUCM is 192.168.200.53
my GW Config is :
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
codec preference 4 g729br8
voice translation-rule 3
rule 1 /^9/ /\1/
voice translation-rule 4
rule 4 /^22217/ /7/
rule 5 /^2217/ /7/
rule 6 /^022217/ /7/
rule 7 /^0122217/ /7/
voice translation-rule 5
rule 1 /^5/ /905/
rule 2 /^1/ /901/
rule 3 /^2/ /902/
rule 4 /^3/ /903/
rule 5 /^4/ /904/
rule 6 /^6/ /906/
rule 7 /^7/ /907/
rule 8 /^8/ /908/
rule 10 /^00/ /900/
rule 11 /'+'/ /900/
voice translation-profile OUT
translate called 3
voice translation-profile REDIAL
translate calling 5
voice translation-profile SIP-NEW
translate called 4
application
service mva http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
service ccm http://192.168.200.53:8080/ccmivr/pages/IVRMainpage.vxml
license udi pid CISCO2921/K9 sn FCZ164960G0
hw-module pvdm 0/0
hw-module pvdm 0/1
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
ip address 192.168.200.86 255.255.255.0
duplex auto
speed auto
interface GigabitEthernet0/1
ip address 172.29.7.94 255.255.255.252
duplex auto
speed auto
ip http server
ip http access-class 23
ip http authentication local
no ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip route 0.0.0.0 0.0.0.0 192.168.200.1
ip route 10.208.9.0 255.255.255.0 172.29.7.93
access-list 23 permit 10.10.10.0 0.0.0.7
control-plane
mgcp profile default
sccp local GigabitEthernet0/0
sccp ccm 192.168.200.53 identifier 1 priority 1 version 7.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 2 register NAGHI-MTP
dspfarm profile 2 mtp
codec g711alaw
maximum sessions hardware 25
associate application SCCP
dial-peer voice 802 voip
description ** SIP TO STC **
translation-profile outgoing OUT
destination-pattern 9T
session protocol sipv2
session target ipv4:10.208.9.69:5060
session transport udp
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay sip-notify rtp-nte sip-kpml
no vad
dial-peer voice 811 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 022217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 812 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 22217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 813 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 2217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 814 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 022217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 815 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 22217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 816 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 2217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 817 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.53
incoming called-number 0122217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
dial-peer voice 818 voip
description ** SIP INCOMING FROM STC **
translation-profile incoming SIP-NEW
translation-profile outgoing REDIAL
preference 1
destination-pattern 7...
session protocol sipv2
session target ipv4:192.168.200.63
incoming called-number 0122217...$
dtmf-relay sip-notify rtp-nte sip-kpml
codec g711alaw
Please i need ur help ASAP

Sip options ping cucm

Hi -
I have a strange problem and I'm looking for ideas. I have a version 9.1.2 CUCM cluster with two nodes. I am setting up two SIP trunks to our CUSP and in CUSP I have configured the CAC to send an Options ping to each CM server. The subscriber is sending a 200 OK reply to the Options ping but the publisher is not.
I set up a packet capture on both servers and I can see the Options ping coming into both. I see the 200 OK from the subscriber but the publisher does not reply. Also, in the RTMT CM logs I can see the Options ping on the subscriber but the publisher log does not show any SIP requests.
These are both HP servers - not VM
I'm unclear on what services or profiles etc. might be missing or misconfiguration of one server over another. I have restarted the publisher as well as all SIP trunks and SIP profiles.
Thanks,
Les

I attached the RTMT output. You will see that there are no SIP messages in this log. I also took a screen shot of the wireshark output from the CLI packet capture. If you look in the RTMT log for the corresponding time (the wireshark is EDT and the RTMT is CDT so RTMT is an hour difference) you will see that there are no messages (15:34:14:613 or 14:34:14:613 in RTMT). I also grabbed a screen shot of the wireshark for the subscriber - you can see the 200 OK reply.
Regards,
Les
Publisher - wireshark output

ASA 5540 8.4 and Voice Gateway (SIP)

have ASA FW with 8.4(2) : inside is connected to lan , DMZ is having SIP Gateway router , i have configured the Firewall will required ports and enabled sip inspection under policy-map but no VOICE traffic is passing through the firewall , iam using static nat
there are ports opened from both side ASA (dmz-inside and other way)
object-group service TCP tcp
port-object eq sip
port-object range 2000 2443
port-object eq h323
port-object eq rtsp
port-object eq 5061
port-object eq 50693
port-object eq 16341
object-group service UDP udp
port-object eq sip
port-object range 16384 32767
port-object eq tftp
port-object eq ntp
port-object eq 1718
port-object eq 1719
port-object eq 5061
port-object eq 57280
Following is inspection
policy-map global_policy
class inspection_default
inspect ftp
inspect h323 h225
inspect h323 ras
inspect rsh
inspect rtsp
inspect skinny
inspect sunrpc
inspect xdmcp
inspect sip
inspect netbios
inspect tftp
inspect ip-options
inspect dns
service-policy global_policy global
phones are not getting registered .....how about the NAT it should be static between DMZ to inside or No NAT statement
any sugesstions would be helpfull

Although it is a document about ASA on another version it explains how SIP inspection works on the ASA. For you to understand this document you need to understand the device that you are configuring behind the ASA and how SIP works in general.
http://www.cisco.com/en/US/products/ps6120/products_configuration_example09186a008081042c.shtml
Send a copy of the complete configuration and show service-policy

Can't establish a Voice gateway (cisco 2911) using SIP with CUCM 9.1

I have configured a Cisco 2911 as a Voice Gateway using SIP (the configuration is attached), but unfortunately can't establish a test call to a phone (CUPC 8.6 SCCP) using csim start. I have done logging the ccsip debug and ccapi debug and attached them. Could anyone help me to solve this problem?

I just did some research on my end and csim is not supported for SIP. The Invite will never be created and sent to the CUCM to initate the call. It disconnects in the router itself with normal cause.
*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Error/sipSPIOutgoingCallSDP:
Could not create source SDP for Outgoing Call
*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Error/sipSPICreateOutboundSDP:
Error in creating an SDP for the outbound call - Check for supported codecs
*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Error/preprocessSetup:
Error during outbound SDP creation
*Apr 18 08:58:48.086: //40/7D08458F8077/SIP/Info/sipSPIInitiateDisconnect: Initiate call disconnect(16) for outgoing call
Please use an actual call to test your dial-peer and integration with call manager. csim will not work.
Hantale
Sree

Lync Federation - Accept SIP Reverse Negotiation (SIP Invite without SDP)

Hello,
Recently I tested a SIP Federation trunk between Lync Server 2013 and non-Lync Client.
In this scenario the Lync Client 2013 support SIP Reverse Negotiation, by other words if SIP Invite without SDP it's sent to Lync Client 2013 it will be accepted by any configuration option?
With the default settings seams that it's not supported with error reason "Error parsing body"
Trace-Correlation-Id: 3549384327
Instance-Id: 4C9
Direction: outgoing
Peer: lynctest.domain.com:2138
Message-Type: response
Start-Line: SIP/2.0 488 Not Acceptable Here
From: "User4" <sip:[email protected]>;tag=3794445243
To: <sip:[email protected]>;epid=abad235729;tag=a130a7e357
Call-ID: [email protected]
CSeq: 12784624 INVITE
Via: SIP/2.0/TLS 172.16.3.51:5065;branch=z9hG4bK-5765F571;rport;alias;received=172.16.3.51;ms-received-port=2138;ms-received-cid=1200
Content-Length: 0
ms-client-diagnostics: 52009;reason="Error parsing body"
Regards,
Claudio

Hello All,
After some analysis I got the following conclusions.
Lync PC Client doesn't accept initial Invite without SDP ( Delayed Offer ).
However our goal was to test the SIP Reverse Media Negotiation mechanism, so we sent initially a dummy SDP for the initial invite and after the connect send a SIP INVITE without SDP and for my surprise the Lync Client accepted and sent his own SDP on the
200 OK and we sent the new SDP offer in the ACK.
However the result was no Audio, and Lync Client kept sending the Audio to the initial INVITE SDP and ignored the new SDP offered in the ACK message.
So my conclusion it's that LYNC Client doesn't support SIP Reverse Media Negotiation (Delayed Offer) at all since it ignores the new SDP offered in the ACK message for the mid call media renegotiation attempt with SIP INVITE without SDP.
Traces:
INVITE sip:172.16.1.87:64425;transport=tls;ms-opaque=a3edae884b;ms-received-cid=1F9C00;grid SIP/2.0
Record-Route: <sip:LYNC2013-FE.domain.sifi:5061;transport=tls;opaque=state:F:Ci.R1f9c00;lr;ms-route-sig=aaCRxLomQ6J6ATKjSZx4vJQ22miSAfUAExqMcDvEWyHdss4x_99VHTLQAA>;tag=C161B833E3EAA57C26010E775AC607C8
Via: SIP/2.0/TLS 172.16.0.37:5061;branch=z9hG4bK97FD40ED.FD1FE32C7CB76CCD;branched=FALSE;ms-internal-info="bb0yvN-Txta-aXcfTMPmVSdyK0kBz7b-pamgWfOIbn8vks4x_9o9kUwQAA"
Via: SIP/2.0/TLS 172.16.13.192:5065;branch=z9hG4bK-57656CED;rport;alias;received=172.16.13.192;ms-received-port=2051;ms-received-cid=1D0800
Proxy-Authentication-Info: Kerberos qop="auth", opaque="D83CD7C8", srand="F5054EF3", snum="104", rspauth="040401ffffffffff0000000000000000e9693240576b479326af5617", targetname="sip/LYNC2013-FE.domain.sifi",
realm="SIP Communications Service", version=4
Max-Forwards: 56
From: "" <sip:[email protected]>;tag=3691888833
To: <sip:[email protected]>;epid=8a34f77d58;tag=4066ac742a
Call-ID: [email protected]
CSeq: 12046301 INVITE
Contact: <sip:[email protected]:5065;transport=TLS>
Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,PRACK,OPTIONS,BYE,CANCEL,REFER,INFO,UPDATE,PUBLISH
Content-Length: 0
Require: 100rel
Supported: 100rel,replaces,privacy,timer,from-change,histinfo,answermode
User-Agent: (Virtual Appliance)
P-Asserted-Identity: "" <sip:[email protected]>
Session-Expires: 720;refresher=uac
P-Sig-Options: Sending-Complete
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 172.16.0.37:5061;branch=z9hG4bK97FD40ED.FD1FE32C7CB76CCD;branched=FALSE;ms-internal-info="bb0yvN-Txta-aXcfTMPmVSdyK0kBz7b-pamgWfOIbn8vks4x_9o9kUwQAA"
Via: SIP/2.0/TLS 172.16.13.192:5065;branch=z9hG4bK-57656CED;rport;alias;received=172.16.13.192;ms-received-port=2051;ms-received-cid=1D0800
From: "" <sip:[email protected]>;tag=3691888833
To: <sip:[email protected]>;epid=8a34f77d58;tag=4066ac742a
Call-ID: [email protected]
CSeq: 12046301 INVITE
User-Agent: UCCAPI/15.0.4701.1000 OC/15.0.4701.1000 (Microsoft Lync)
Proxy-Authorization: Kerberos qop="auth", realm="SIP Communications Service", opaque="D83CD7C8", targetname="sip/LYNC2013-FE.domain.sifi", crand="785246a1", cnum="92", response="040400ffffffffff000000000000000000b60640ac2c60c49bc1b427"
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/TLS 172.16.0.37:5061;branch=z9hG4bK97FD40ED.FD1FE32C7CB76CCD;branched=FALSE;ms-internal-info="bb0yvN-Txta-aXcfTMPmVSdyK0kBz7b-pamgWfOIbn8vks4x_9o9kUwQAA"
Via: SIP/2.0/TLS 172.16.13.192:5065;branch=z9hG4bK-57656CED;rport;alias;received=172.16.13.192;ms-received-port=2051;ms-received-cid=1D0800
From: "" <sip:[email protected]>;tag=3691888833
To: <sip:[email protected]>;epid=8a34f77d58;tag=4066ac742a
Call-ID: [email protected]
CSeq: 12046301 INVITE
Record-Route: <sip:LYNC2013-FE.domain.sifi:5061;transport=tls;opaque=state:F:Ci.R1f9c00;lr;ms-route-sig=aaCRxLomQ6J6ATKjSZx4vJQ22miSAfUAExqMcDvEWyHdss4x_99VHTLQAA>;tag=C161B833E3EAA57C26010E775AC607C8
Contact: <sip:[email protected];opaque=user:epid:wc5Y6-kDo16CxuVbyxqk9gAA;gruu>
User-Agent: UCCAPI/15.0.4701.1000 OC/15.0.4701.1000 (Microsoft Lync)
Supported: histinfo
Supported: ms-safe-transfer
Supported: ms-dialog-route-set-update
Proxy-Authorization: Kerberos qop="auth", realm="SIP Communications Service", opaque="D83CD7C8", targetname="sip/LYNC2013-FE.domain.sifi", crand="e903d142", cnum="93", response="040400ffffffffff0000000000000000dbe0e9524a1031ef81a19d2f"
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 0 1 IN IP4 172.16.1.87
s=session
c=IN IP4 172.16.1.87
b=CT:99980
t=0 0
m=audio 12530 RTP/SAVP 8 0 13 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:mIMHiJBpn4ZRZfg2VXYSTdQfS4wyJ0x57QQ0q4kU|2^31
a=maxptime:200
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
ACK sip:172.16.1.87:64425;transport=tls;ms-opaque=a3edae884b;ms-received-cid=1F9C00;grid SIP/2.0
Via: SIP/2.0/TLS 172.16.0.37:5061;branch=z9hG4bK301D467E.2E943CC97CBC4CCD;branched=FALSE
Via: SIP/2.0/TLS 172.16.13.192:5065;branch=z9hG4bK-57656CEE;rport;received=172.16.13.192;ms-received-port=2051;ms-received-cid=1D0800
Proxy-Authentication-Info: Kerberos qop="auth", opaque="D83CD7C8", srand="B8AB5336", snum="105", rspauth="040401ffffffffff0000000000000000de85d6c7415302c9b7535777", targetname="sip/LYNC2013-FE.domain.sifi",
realm="SIP Communications Service", version=4
Max-Forwards: 69
From: "" <sip:[email protected]>;tag=3691888833
To: <sip:[email protected]>;epid=8a34f77d58;tag=4066ac742a
Call-ID: [email protected]
CSeq: 12046301 ACK
Contact: <sip:[email protected]:5065;transport=TLS>
Content-Length: 326
Content-Type: application/sdp
v=0
o=- 262 2 IN IP4 172.16.13.192
s=session
t=0 0
m=audio 16392 RTP/SAVP 8 101 13
c=IN IP4 172.16.13.191
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:X0rDwl9KxCJfSsRaX0rEkl9KxNJfSsUCX0rFOtIK|2^31

Cucm 10.1version - training videos and hand guides on understanding voice gateway h323 and SIP? thanks

cucm 10.1version - any free training videos and hand guides on understanding voice gateway h323 and SIP and how to configure one? thanks

Learncisco gives a very good introduction to CUCM - I recommend you start there.

**ERROR**: CCPMSG_InCall: ChannelID IE invalid 100(0x64) in SIP voice gateway

Hi all,
I am configuring a SIP voice gateway. After finsihing the config, the outbound calls are working properly. But I met some issue for inbound calls.
There is no debug log for show ccsip message and show voice dialpeer command. Only q931 has the debug log.
Can any one have any idea for this issue?
below is the debug shows for show isdn q931.
   Jun 1 05:27:49.435: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8 callref = 0x03A3
Sending Complete
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xE1858384
Preferred, Interface 5, Channel 4
Calling Party Number i = 0x2181, '18600586101'
Plan:ISDN, Type:National
Called Party Number i = 0xC1, '82197910'
Plan:ISDN, Type:Subscriber(local)
High Layer Compat i = 0x9181
High Layer Compat i = 0x9181
Jun 1 05:27:49.439: ISDN Se0/0/0:15 Q931: Received SETUP callref = 0x83A3 callID = 0x0004 switch = primary-net5 interface = User
d8w-sr-2811f20a-vpn#
Jun 1 05:27:49.439: ISDN Se0/0/0:15 **ERROR**: CCPMSG_InCall: ChannelID IE invalid 100(0x64) rejecting call
Jun 1 05:27:49.443: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x83A3
Cause i = 0x80E418 - Invalid information element contents
下午01:28:17: Zhuliang: Jun 1 05:27:49.439: ISDN Se0/0/0:15 **ERROR**: CCPMSG_InCall: ChannelID IE invalid 100(0x64) rejecting call
Jun 1 05:27:49.443: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x83A3
Cause i = 0x80E418 - Invalid information element contents

Channel ID i = 0xE1858384 Preferred, Interface 5, Channel 4 Jun 1 05:27:49.443: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x83A3 Cause i = 0x80E418 - Invalid information element contents下午01:28:17: Zhuliang: Jun 1 05:27:49.439: ISDN Se0/0/0:15 **ERROR**: CCPMSG_InCall: ChannelID IE invalid 100(0x64) rejecting callJun 1 05:27:49.443: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x83A3 Cause i = 0x80E418 - Invalid information element contents
To what are you connecting? Call comes with explicit interface indicator IE, that is not normal for ISDN E1.

Call Manager register fxs port with voice gateway- problem

I have a CUCM 6 and a Voice Gateway V224. I've configured the voice gateway's voice FXS ports as MGCP.
I have a Voip connected and registered to the CUCM and a Pots phone connected to the Voice Gateway.
If i dial from the Voip to the Pots phone it rings. The problem is that i cannot ring from the Pots to the Voip phone.
I have no dial tone.
If i write no shut down on the voice port i have a tone. If i configure mgcp on the voice port i have a busy ringtone.
I've entered no mgcp and mgcp commands and i've reset the voice gateway.
How can i call from the pots to the voip phone?
The ios version on the voice gateway is Version 12.4(22)T4.
Here is an outghtput from the Voice gateway.
ccm-manager mgcp
ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 10.1.1.33
ccm-manager config
mgcp
mgcp call-agent CCMIOSS 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp validate domain-name
mgcp rtp payload-type g726r16 static
mgcp profile default
timeout tone busy 600
timeout tone dial 600
dial-peer voice 999223 pots
service mgcpapp
port 2/23
dial-peer voice 999222 pots
service mgcpapp
port 2/22
dial-peer voice 999888 pots
service mgcpapp
port 2/23
The CUCM 6 is registered with the voice gateway.

Is your campaign using CPA? If so, what's the behavior if CPA is not enabled?
I think the best thing to do is to run a trace...
Call Manager > Cisco Unified Serviceability > Trace > Configurations
Select a CUCM server - any subscriber would work.
Service Group - CM Services
Cisco CallManager (Inactive)
Enable SIP Stack Trace and apply to all nodes. Download and install RTMT
Make a bunch of outbound tests and then open RTMT > Trace & Log Central > Collect Files > Check "All Servers" for Cisco CallManager > Next > Next > Relative Range if you made the test calls within the last X minutes, otherwise you can set a From and To datetime. Click Finish and go through your SDL logs and see what errors you find and post them here.
Also, make sure your phone is in the correct CSS in Call Manager

ISR as CUBE and Voice Gateway

Can I set an ISR 2951 as CUBE to receive SIP trunks and configure the same box as voice gateway to deliver TDM E1 voice channels to an enterprise PBx?
Thanks
Sent from Cisco Technical Support iPhone App

Yes You can...
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How to make CUCM as a TFTP server , then copy files to Voice Gateway ?

how to make CUCM as a TFTP server , then copy files to Voice Gateway ? anyone knows?

Hi,
Please check the following link
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/6_1_1/ccmsys/cucm-accm-611/a02tftp.html#wp1023004
Understanding How Devices Access the TFTP Server
You can enable the IP phones and gateways to discover the TFTP server IP address in one or more of the following ways, depending on the device type:
•Gateways and phones can use DHCP custom option 150.
Cisco recommends this method. With this method, you configure the TFTP server IP address as the option value.
•Gateways and phones can use DHCP option 066.
You may configure either the host name or IP address of the TFTP server as the option value.
•Gateways and phones can query CiscoCM1.
Ensure the Domain Name System (DNS) can resolve this name to the IP address of the TFTP server. Cisco does not recommend this option because it does not scale.
•You can configure phones with the IP address of the TFTP server. If DHCP is enabled on the phone, you can still configure an alternate TFTP server IP address locally on the phone that will override the TFTP address that was obtained through DHCP.
•Gateways and phones also accept the DHCP Optional Server Name (sname) parameter.
•The phone or gateway can use the value of Next-Server in the boot processes (siaddr).
Devices save the TFTP server address in nonvolatile memory. If one of the preceding methods was available at least once, but is not currently available, the device uses the address that is saved in memory.
You can configure the TFTP service on the first node or a subsequent node, but usually you should configure it on the first node. For small systems, the TFTP server can coexist with a Cisco Unified Communications Manager on the same server.
HTH
Manish

Changing CAS e&m-wink-start to a PRI on voice gateway

Cisco IOS Software, C3900 Software (C3900-UNIVERSALK9-M), Version 15.1(4)M4, RELEASE SOFTWARE (fc1)
I want to change a CAS e&m-wink-start to a PRI on controller T1 0/1/0 using mgcp below. I am posting the new configuration below the current e&m-wink-start configuration on the voice gateway router.
Current
controller T1 0/1/0
cablelength long 0db
ds0-group 1 timeslots 1-24 type e&m-wink-start
description
New configuration
Router(config)# no contoller T1 0/1/0
Router(config)# no interface Serial0/1/0
Router(config)#controller t1 0/1/0
Router(config-controller)#cablelength long 0db
Router(config-controller)#framing esf
Router(config-controller)#linecode b8zs
Router(config-controller)#clock source line
Router(config-controller)#pri-group timeslots 1-24 service mgcp
Router(config-controller)#description circuit ID
Router(config-if)# interface serial0/1/0
Router(config-if)# no ip address
Router(config-if)# encapsulation hdlc
Router(config-if)# isdn switch-type primary-4ess
Router(config-if)# isdn incoming-voice voice
Router(config-if)# isdn bind-l3 ccm-manager

Hi,
You need additional configs for this to work..You also need to configure the isdn bind on the D-channel not as you have it..
Router(config-if)# interface serial0/1/0:23
Router(config-if)# no ip address
Router(config-if)# encapsulation hdlc
Router(config-if)# isdn switch-type primary-4ess
Router(config-if)# isdn incoming-voice voice
Router(config-if)# isdn bind-l3 ccm-manager
Additional configs....change ip to suit your needs
ccm-manager redundant-host 192.168.103.114
ccm-manager mgcp
ccm-manager music-on-hold bind xxx--put relevant interface
mgcp
mgcp call-agent 192.168.103.115 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp modem passthrough voip codec g711alaw
mgcp modem passthrough voip redundancy
mgcp ip qos dscp af31 media
mgcp ip qos dscp cs3 signaling
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
mgcp default-package dtmf-package
no mgcp package-capability res-package
no mgcp package-capability fxr-package
mgcp tse payload 100
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp rtp payload-type g726r16 static
mgcp bind control source-interfacexx -----------------------put relevant interfcae here
mgcp bind media source-interface xxx--------------------same
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Voice gateway, SIP Options ping without SDP

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