[cisco-voip] SIP gateway config

[Peter Slow]

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
>>>>
>>>>
>>>>
>>>>
>>>>
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>>>
>>>
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>>>
>>
>
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