[cisco-voip] SIP gateway config

Peter Slow peter.slow at gmail.com
Mon Apr 26 18:01:59 EDT 2010


ah, i missed part of your problem description, but the first thing i'd do is
fix your dialplan so that you dont send failed calls to CUCM back to your
telco, which it looks liek you're doing. IT woudl seem that that's whay your
outbound setup message is in this case.

anyway, i see what you're saying with the 401 unauthorized.

coudl we see a debug of the good (outbound) call you mentioned?

-Peter

On Mon, Apr 26, 2010 at 5:58 PM, Brian Schultz <bms314 at gmail.com> wrote:

> ??  Not sure i understand this...
>
>
>
>
> On Mon, Apr 26, 2010 at 4:56 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
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
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>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
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>>>>>
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>>>>>
>>>>>
>>>>
>>>
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>>
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