[cisco-voip] SIP gateway config
Brian Schultz
bms314 at gmail.com
Mon Apr 26 18:38:51 EDT 2010
I'm going to start a new post since this is a different problem...
Thanks,
Brian
On Mon, Apr 26, 2010 at 5:16 PM, Brian Schultz <bms314 at gmail.com> wrote:
> That was it, thanks Justin. Somehow digest authentication was checked on
> the SIP security profile.
>
> Now, I have a new problem. So far this problem has been intermittent. On
> some calls, the SIP setup messages don't start until 7 seconds after the
> call hits the gateway. I see the ISDN debug and hear a message from the
> provider that the call couldn't be completed as dialed. Then ISDN releases
> the call and I immediately see a SIP invite and my phone starts to ring and
> completes as normal. I've reproduced it several times already. Very
> strange. Here is an example:
>
>
> 002234: Apr 26 17:13:54 Central: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd =
> 8 callref = 0x8001
>
> Channel ID i = 0xA98381
> Exclusive, Channel 1
> 002235: Apr 26 17:13:54 Central: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8
> callref = 0x00A0
>
> Bearer Capability i = 0x8090A2
> Standard = CCITT
> Transfer Capability = Speech
> Transfer Mode = Circuit
> Transfer Rate = 64 kbit/s
> Channel ID i = 0xA98397
> Exclusive, Channel 23
> Calling Party Number i = 0x2183, '9528183360'
> Plan:ISDN, Type:National
> Called Party Number i = 0xC1, '7715'
> Plan:ISDN, Type:Subscriber(local)
> 002236: Apr 26 17:13:54 Central: ISDN Se0/0/0:23 Q931: RX <- CALL_PROC pd =
> 8 callref = 0x80A0
>
> Channel ID i = 0xA98397
> Exclusive, Channel 23
> UHD.EAST.RTR1#
> 002237: Apr 26 17:13:54 Central: ISDN Se0/0/0:23 Q931: RX <- PROGRESS pd =
> 8 callref = 0x80A0
>
> Cause i = 0x8283 - No route to destination
> Progress Ind i = 0x8288 - In-band info or appropriate now available
> 002238: Apr 26 17:13:54 Central: ISDN Se0/0/0:23 Q931: TX -> ALERTING pd =
> 8 callref = 0x8001
>
> Progress Ind i = 0x8288 - In-band info or appropriate now available
> UHD.EAST.RTR1#
> 002239: Apr 26 17:14:01 Central: ISDN Se0/0/0:23 Q931: RX <- DISCONNECT pd
> = 8 callref = 0x80A0
>
> Cause i = 0x8290 - Normal call clearing
> 002240: Apr 26 17:14:01 Central: ISDN Se0/0/0:23 Q931: TX -> RELEASE pd =
> 8 callref = 0x00A0
> 002241: Apr 26 17:14:01 Central: ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP
> pd = 8 callref = 0x80A0
> 002242: Apr 26 17:14:01 Central: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>
>
> Sent:
> INVITE sip:7715 at 172.21.20.10:5060 SIP/2.0
> Via: SIP/2.0/UDP 172.21.20.254:5060;branch=z9hG4bK5ED1B0D
>
> Remote-Party-ID: <sip:9528183360 at 172.21.20.254<sip%3A9528183360 at 172.21.20.254>
> >;party=calling;screen=yes;privacy=off
> From: <sip:9528183360 at 172.21.20.254 <sip%3A9528183360 at 172.21.20.254>>;tag=3388AB18-55
>
>
> To: <sip:7715 at 172.21.20.10 <sip%3A7715 at 172.21.20.10>>
> Date: Mon, 26 Apr 2010 22:14:01 GMT
> Call-ID: DA74D879-50B711DF-819BFADB-42390B at 172.21.20.254
>
> Supported: 100rel,timer,resource-priority,replaces,sdp-anat
> Min-SE: 1800
> Cisco-Guid: 3595010882-1354174943-2151188547-3779644176
>
> User-Agent: Cisco-SIPGateway/IOS-12.x
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
> NOTIFY, INFO, REGISTER
> CSeq: 101 INVITE
> Max-Forwards: 70
> Timestamp: 1272320041
>
> Contact: <sip:9528183360 at 172.21.20.254:5060>
> Expires: 180
> Allow-Events: telephone-event
> Content-Type: application/sdp
> Content-Disposition: session;handling=required
> Content-Length: 330
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 5784 5366 IN IP4 172.21.20.254
>
> s=SIP Call
> c=IN IP4 172.21.20.254
> t=0 0
> m=audio 26090 RTP/AVP 0 18 9 101
>
> c=IN IP4 172.21.20.254
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:9 G722/8000
> a=fmtp:9 bitrate=64
>
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> 002243: Apr 26 17:14:01 Central: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>
>
> Received:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 172.21.20.254:5060;branch=z9hG4bK5ED1B0D
> From: <sip:9528183360 at 172.21.20.254 <sip%3A9528183360 at 172.21.20.254>>;tag=3388AB18-55
>
>
> To: <sip:7715 at 172.21.20.10 <sip%3A7715 at 172.21.20.10>>
> Date: Mon, 26 Apr 2010 22:14:20 GMT
> Call-ID: DA74D879-50B711DF-819BFADB-42390B at 172.21.20.254
>
> CSeq: 101 INVITE
> Allow-Events: presence
> Content-Length: 0
>
> 002244: Apr 26 17:14:01 Central: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Received:
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 172.21.20.254:5060;branch=z9hG4bK5ED1B0D
> From: <sip:9528183360 at 172.21.20.254 <sip%3A9528183360 at 172.21.20.254>
> >;tag=3388AB18-55
> To: <sip:7715 at 172.21.20.10 <sip%3A7715 at 172.21.20.10>
> >;tag=fc6aef00-fcb6-4e1c-a8c0-e38242535154-20774950
> Date: Mon, 26 Apr 2010 22:14:20 GMT
> Call-ID: DA74D879-50B711DF-819BFADB-42390B at 172.21.20.254
> CSeq: 101 INVITE
> Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> SUBSCRIBE, NOTIFY
> Allow-Events: presence
> Contact: <sip:7715 at 172.21.20.10:5060>
> Supported: X-cisco-srtp-fallback
> UHD.EAST.RTR1#Supported: Geolocation
> P-Asserted-Identity: "Phenomenal Networks" <sip:7715 at 172.21.20.10<sip%3A7715 at 172.21.20.10>
> >
> Remote-Party-ID: "Phenomenal Networks" <sip:7715 at 172.21.20.10<sip%3A7715 at 172.21.20.10>
> >;party=called;screen=yes;privacy=off
> Content-Length: 0
>
> UHD.EAST.RTR1#
> 002245: Apr 26 17:14:06 Central: ISDN Se0/0/0:23 Q931: RX <- DISCONNECT pd
> = 8 callref = 0x0001
>
> Cause i = 0x8290 - Normal call clearing
> 002246: Apr 26 17:14:06 Central: ISDN Se0/0/0:23 Q931: TX -> RELEASE pd =
> 8 callref = 0x8001
> 002247: Apr 26 17:14:06 Central: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Sent:
> CANCEL sip:7715 at 172.21.20.10:5060 SIP/2.0
> Via: SIP/2.0/UDP 172.21.20.254:5060;branch=z9hG4bK5ED1B0D
> From: <sip:9528183360 at 172.21.20.254 <sip%3A9528183360 at 172.21.20.254>>;tag=3388AB18-55
>
>
> To: <sip:7715 at 172.21.20.10 <sip%3A7715 at 172.21.20.10>>
> Date: Mon, 26 Apr 2010 22:14:01 GMT
> Call-ID: DA74D879-50B711DF-819BFADB-42390B at 172.21.20.254
> CSeq: 101 CANCEL
> Max-Forwards: 70
> Timestamp: 1272320046
> Reason: Q.850;cause=16
> Content-Length: 0
>
> 002248: Apr 26 17:14:06 Central: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
> Received:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 172.21.20.254:5060;branch=z9hG4bK5ED1B0D
> From: <sip:9528183360 at 172.21.20.254 <sip%3A9528183360 at 172.21.20.254>>;tag=3388AB18-55
>
>
> To: <sip:7715 at 172.21.20.10 <sip%3A7715 at 172.21.20.10>>
> Date: Mon, 26 Apr 2010 22:14:25 GMT
> Call-ID: DA74D879-50B711DF-819BFADB-42390B at 172.21.20.254
> CSeq: 101 CANCEL
> Content-Length: 0
>
>
> On Mon, Apr 26, 2010 at 5:01 PM, Justin Steinberg <jsteinberg at gmail.com>wrote:
>
>> brian,
>>
>> The CM trunk is challenging the IOS for digest authentication. take a
>> look at the SIP security profile you've assigned to the CM SIP trunk and
>> configure it for a profile that doesn't require digest authentication.
>>
>> On Mon, Apr 26, 2010 at 5:56 PM, Peter Slow <peter.slow at gmail.com>wrote:
>>
>>> Brian,
>>>
>>>
>>> 002008: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: TX -> SETUP pd =
>>> 8 callref = 0x0097
>>> <snip>
>>>
>>> Called Party Number i = 0xC1, '7715'
>>> Plan:ISDN, Type:Subscriber(local)
>>>
>>>
>>> -Pete
>>>
>>>
>>> On Mon, Apr 26, 2010 at 5:35 PM, Brian Schultz <bms314 at gmail.com> wrote:
>>>
>>>> I only have MTP configured on the CUCM server as part of the Device
>>>> Pool. Do I need separate MTP configured on the router?
>>>>
>>>> Here is my debug. 7715 is the DID which I have configured on my soft
>>>> phone. I used RDM to create base config with SIP trunks to gateways. Trunk
>>>> is in the same CSS and Device Pool as the phone.
>>>>
>>>>
>>>> 002000: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: RX <- SETUP pd =
>>>> 8 callref = 0x0001
>>>> Bearer Capability i = 0x8090A2
>>>> Standard = CCITT
>>>> Transfer Capability = Speech
>>>> Transfer Mode = Circuit
>>>> Transfer Rate = 64 kbit/s
>>>> Channel ID i = 0xA98381
>>>> Exclusive, Channel 1
>>>> Calling Party Number i = 0x2183, '9528183360'
>>>> Plan:ISDN, Type:National
>>>> Called Party Number i = 0xC1, '7715'
>>>> Plan:ISDN, Type:Subscriber(local)
>>>> 002001: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: Received SETUP
>>>> callref = 0x8001 callID = 0x0042 switch = primary-5ess interface = User
>>>> 002002: Apr 26 16:31:12 Central:
>>>> //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>>> Sent:
>>>> INVITE sip:7715 at 172.21.20.10:5060 SIP/2.0
>>>> Via: SIP/2.0/UDP 172.21.20.254:5060;branch=z9hG4bK5DDED4
>>>> Remote-Party-ID: <sip:9528183360 at 172.21.20.254<sip%3A9528183360 at 172.21.20.254>
>>>> >;party=calling;screen=yes;privacy=off
>>>> From: <sip:9528183360 at 172.21.20.254 <sip%3A9528183360 at 172.21.20.254>
>>>> >;tag=33617790-169F
>>>> To: <sip:7715 at 172.21.20.10 <sip%3A7715 at 172.21.20.10>>
>>>> Date: Mon, 26 Apr 2010 21:31:12 GMT
>>>> Call-ID: DF28E48D-50B111DF-814BFADB-42390B at 172.21.20.254
>>>> Supported: 100rel,timer,resource-priority,replaces,sdp-anat
>>>> Min-SE: 1800
>>>> Cisco-Guid: 3743959142-1353781727-2150402115-3779644176
>>>> User-Agent: Cisco-SIPGateway/IOS-12.x
>>>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>>>> SUBSCRIBE, NOTIFY, INFO, REGISTER
>>>> CSeq: 101 INVITE
>>>> Max-Forwards: 70
>>>> Timestamp: 1272317472
>>>> Contact: <sip:9528183360 at 172.21.20.254:5060>
>>>> Expires: 180
>>>> Allow-Events: telephone-event
>>>> Content-Type: application/sdp
>>>> Content-Disposition: session;handling=required
>>>> Content-Length: 285
>>>> v=0
>>>> o=CiscoSystemsSIP-GW-UserAgent 9351 4649 IN IP4 172.21.20.254
>>>> s=SIP Call
>>>> c=IN IP4 172.21.20.254
>>>> t=0 0
>>>> m=audio 25022 RTP/AVP 0 18 101
>>>> c=IN IP4 172.21.20.254
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:18 G729/8000
>>>> a=fmtp:18 annexb=no
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> 002003: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC
>>>> pd = 8 callref = 0x8001
>>>> Channel ID i = 0xA98381
>>>> Exclusive, Channel 1
>>>> 002004: Apr 26 16:31:12 Central:
>>>> //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>>> Received:
>>>> SIP/2.0 100 Trying
>>>> Via: SIP/2.0/UDP 172.21.20.254:5060;branch=z9hG4bK5DDED4
>>>> From: <sip:9528183360 at 172.21.20.254 <sip%3A9528183360 at 172.21.20.254>
>>>> >;tag=33617790-169F
>>>> To: <sip:7715 at 172.21.20.10 <sip%3A7715 at 172.21.20.10>>
>>>> Date: Mon, 26 Apr 2010 21:31:31 GMT
>>>> Call-ID: DF28E48D-50B111DF-814BFADB-42390B at 172.21.20.254
>>>> CSeq: 101 INVITE
>>>> Allow-Events: presence
>>>> Content-Length: 0
>>>>
>>>> 002005: Apr 26 16:31:12 Central:
>>>> //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>>> Received:
>>>> SIP/2.0 401 Unauthorized
>>>> Via: SIP/2.0/UDP 172.21.20.254:5060;branch=z9hG4bK5DDED4
>>>> From: <sip:9528183360 at 172.21.20.254 <sip%3A9528183360 at 172.21.20.254>
>>>> >;tag=33617790-169F
>>>> To: <sip:7715 at 172.21.20.10 <sip%3A7715 at 172.21.20.10>>;tag=205608936
>>>> Date: Mon, 26 Apr 2010 21:31:31 GMT
>>>> Call-ID: DF28E48D-50B111DF-814BFADB-42390B at 172.21.20.254
>>>> CSeq: 101 INVITE
>>>> Allow-Events: presence
>>>> WWW-Authenticate: Digest realm="StandAloneCluster",
>>>> nonce="oQbua7BD9UUXn7PtEyhEPuxTU4a5UWsT", algorithm=MD5
>>>> Content-Length: 0
>>>>
>>>> 002006: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: Applying typeplan
>>>> for sw-type 0x3 is 0x2 0x1, Calling num 9528183360
>>>> 002007: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: Sending SETUP
>>>> callref = 0x0097 callID = 0x8018 switch = primary-5ess interface = User
>>>> 002008: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: TX -> SETUP pd =
>>>> 8 callref = 0x0097
>>>> Bearer Capability i = 0x8090A2
>>>> Standard = CCITT
>>>> Transfer Capability = Speech
>>>> Transfer Mode = Circuit
>>>> Transfer Rate = 64 kbit/s
>>>> Channel ID i = 0xA98397
>>>> Exclusive, Channel 23
>>>> Calling Party Number i = 0x2183, '9528183360'
>>>> Plan:ISDN, Type:National
>>>> Called Party Number i = 0xC1, '7715'
>>>> Plan:ISDN, Type:Subscriber(local)
>>>> 002009: Apr 26 16:31:12 Central:
>>>> //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
>>>> Sent:
>>>> ACK sip:7715 at 172.21.20.10:5060 SIP/2.0
>>>> Via: SIP/2.0/UDP 172.21.20.254:5060;branch=z9hG4bK5DDED4
>>>> From: <sip:9528183360 at 172.21.20.254 <sip%3A9528183360 at 172.21.20.254>
>>>> >;tag=33617790-169F
>>>> To: <sip:7715 at 172.21.20.10 <sip%3A7715 at 172.21.20.10>>;tag=205608936
>>>> Date: Mon, 26 Apr 2010 21:31:12 GMT
>>>> Call-ID: DF28E48D-50B111DF-814BFADB-42390B at 172.21.20.254
>>>> Max-Forwards: 70
>>>> CSeq: 101 ACK
>>>> Allow-Events: telephone-event
>>>> Content-Length: 0
>>>>
>>>> 002010: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: RX <- CALL_PROC
>>>> pd = 8 callref = 0x8097
>>>> Channel ID i = 0xA98397
>>>> Exclusive, Channel 23
>>>> UHD.EAST.RTR1#
>>>> 002011: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: RX <- PROGRESS pd
>>>> = 8 callref = 0x8097
>>>> Cause i = 0x8283 - No route to destination
>>>> Progress Ind i = 0x8288 - In-band info or appropriate now
>>>> available
>>>> 002012: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: TX -> ALERTING pd
>>>> = 8 callref = 0x8001
>>>> Progress Ind i = 0x8288 - In-band info or appropriate now
>>>> available
>>>> UHD.EAST.RTR1#
>>>> 002013: Apr 26 16:31:18 Central: ISDN Se0/0/0:23 Q931: RX <- DISCONNECT
>>>> pd = 8 callref = 0x0001
>>>> Cause i = 0x8290 - Normal call clearing
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Mon, Apr 26, 2010 at 4:06 PM, miken miken <miken at sisna.com> wrote:
>>>>
>>>>> MTP configured and box checked mandatory in SIP trunk configuration on
>>>>> CUCM?
>>>>>
>>>>> Thanks
>>>>> MikeN
>>>>>
>>>>> On Mon, Apr 26, 2010 at 3:00 PM, Brian Schultz <bms314 at gmail.com>wrote:
>>>>>
>>>>>> Yep, have that already. Gig0/1.110 has the IP address configured
>>>>>> on the SIP trunk in CUCM.
>>>>>>
>>>>>> voice service voip
>>>>>> sip
>>>>>> bind control source-interface GigabitEthernet0/1.110
>>>>>> bind media source-interface GigabitEthernet0/1.110
>>>>>>
>>>>>> I also have the following:
>>>>>>
>>>>>> voice class codec 1
>>>>>> codec preference 1 g711ulaw
>>>>>> codec preference 2 g729r8
>>>>>> dial-peer voice 100 voip
>>>>>> destination-pattern ....
>>>>>> session protocol sipv2
>>>>>> session target ipv4:172.21.20.10
>>>>>> voice-class codec 1
>>>>>> dtmf-relay rtp-nte
>>>>>> no vad
>>>>>> dial-peer voice 1 pots
>>>>>> incoming called-number .
>>>>>> direct-inward-dial
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Mon, Apr 26, 2010 at 3:58 PM, Ahmed Elnagar <
>>>>>> ahmed_elnagar at rayacorp.com> wrote:
>>>>>>
>>>>>>> You have to bind signal and media traffic out of the router to the
>>>>>>> CUCM with the IP address you have configured on CUCM “by default CUCM reject
>>>>>>> calls with source address other than the configured”
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Try the below on the gateway:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> voice service voip
>>>>>>>
>>>>>>> sip
>>>>>>>
>>>>>>> bind all source-interface “interface configured on CUCM”
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Best Regards;
>>>>>>>
>>>>>>> Ahmed Elnagar
>>>>>>>
>>>>>>> Senior Network PS Engineer
>>>>>>>
>>>>>>> Mob: +2019-0016211
>>>>>>>
>>>>>>> [image: ccie_voice_large.gif][image: ccvp_voice_large.gif]
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> *From:* cisco-voip-bounces at puck.nether.net [mailto:
>>>>>>> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Brian Schultz
>>>>>>> *Sent:* Monday, April 26, 2010 10:34 PM
>>>>>>> *To:* cisco-voip at puck.nether.net
>>>>>>> *Subject:* [cisco-voip] SIP gateway config
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Does anyone happen to have an example SIP gateway config for an ISR?
>>>>>>> CUCM 8.0(2) with a SIP trunk to a 2921 gateway (15.0.M1.12) with a standard
>>>>>>> PRI for PSTN access. I have outbound working, but inbound gives a fast busy
>>>>>>> with a 401 Unauthorized in the SIP debug.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Thanks,
>>>>>>>
>>>>>>> Brian
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Disclaimer: NOTICE The information contained in this message is
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>>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> cisco-voip mailing list
>>>>>>
>>>>>> cisco-voip at puck.nether.net
>>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>>
>>>>>>
>>>>>
>>>>
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>>>>
>>>
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