[cisco-voip] SIP gateway config

Justin Steinberg jsteinberg at gmail.com
Mon Apr 26 18:01:30 EDT 2010


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