[cisco-voip] SIP gateway config

Brian Schultz bms314 at gmail.com
Mon Apr 26 17:35:59 EDT 2010


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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