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CME SIP TRUNK WITH SONUS GSX9000 HD NO REGISTER (Received : SIP/2.0 405 Method Not Allowed)

greetings to all..

i had an CME trying to connecto to a SIP trunk with provider with Sonux GSX9000 HD, the telephony provider does not support credential authentication and i can not stablish the session to stay engaged with the provider and receive the calls from PSTN.

the topology is that.

cme-h323-voipgw-sip-trunk.jpg

i'll provide tomorrow a visio topology to show you how is the real network.

Here is the log :

Sent:

REGISTER sip:signal.convergia.com:5060 SIP/2.0

Via: SIP/2.0/UDP 10.255.250.30:5060;branch=z9hG4bK8126F

From: <sip:099........@signal.convergia.com>;tag=705618-2183

To: <sip:099........@signal.convergia.com>

Date: Wed, 15 Jan 2014 02:03:08 GMT

Call-ID: 6F618D1-7CAF11E3-8002808A-B237480

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1389751388

CSeq: 6 REGISTER

Contact: <sip:099........@10.255.250.30:5060>

Expires:  3600

Supported: path

Content-Length: 0

Jan 15 02:03:08.551: //17/000000000000/SIP/Msg/ccsipDisplayMsg:

Received: #SIP/2.0 405 Method Not Allowed

Via: SIP/2.0/UDP 10.255.250.30:5060;branch=z9hG4bK8126F;received=200.27.121.158

From: <sip:099........@signal.convergia.com>;tag=705618-2183

To: <sip:099........@signal.convergia.com>

Call-ID: 6F618D1-7CAF11E3-8002808A-B237480

CSeq: 6 REGISTER

Allow: INVITE,ACK,CANCEL,BYE,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS

Content-Length: 0

Here are the relevant configurations.

voice service voip

ip address trusted list

  ipv4 172.30.253.253 255.255.255.255

  ipv4 200.75.6.66 255.255.255.255

  ipv4 200.75.6.67 255.255.255.255

  ipv4 200.27.121.158 255.255.255.255

dtmf-interworking standard

media transcoder sync-streams

media anti-trombone

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

no supplementary-service sip moved-temporarily

no supplementary-service sip handle-replaces

no fax-relay sg3-to-g3

h323

  ip circuit max-calls 40960

  no h225 timeout keepalive

  no call service stop

  call start slow

  session transport udp

  modem passthrough nse codec g711ulaw

sip

  bind control source-interface FastEthernet0/1

  bind media source-interface FastEthernet0/1

  registrar server expires max 120 min 60

  localhost dns:signal.convergia.com

  no update-callerid

  g729 annexb-all

  no call service stop

!

sip-ua

no remote-party-id

retry invite 10

retry response 10

retry bye 2

retry cancel 2

retry register 10

retry options 5

timers connect 100

timers register 250

registrar 1 dns:signal.convergia.com expires 3600 refresh-ratio 50

sip-server dns:signal.convergia.com

host-registrar

permit hostname dns:media.convergia.com

permit hostname dns:signal.convergia.com

!

dial-peer voice 10000 voip

description CONVERGIA-SIP-OUTGOING

translation-profile outgoing PSTN_Outgoing

destination-pattern .

session protocol sipv2

session target sip-server

voice-class codec 2 

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

no vad

!

dial-peer voice 10001000 pots

description OUTGOING_VOICE_DIAL-PEER.CELL.PSTN

destination-pattern 099........

port 0/2/0

forward-digits 8

!

voice translation-profile PSTN_Outgoing

translate calling 10

translate called 20

!

voice translation-rule 10

rule 1 /^.*/ /227602276/

!

voice translation-rule 20

rule 1 /\(.*\)/ /56227602276/

!

voice class codec 2

codec preference 1 g729r8

!

Hope you could help me .

had a great day . best regards, and rate if you'll find this post useful
  • IP Telephony
5 REPLIES

Re: CME SIP TRUNK WITH SONUS GSX9000 HD NO REGISTER (Received :

Your ISP only allow the following header you will need to remove the register part from the SIP message:

Allow: INVITE,ACK,CANCEL,BYE,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
Content-Length: 0


Sent from Cisco Technical Support iPhone App

Voice CCIE #37771

CME SIP TRUNK WITH SONUS GSX9000 HD NO REGISTER (Received : SIP/

Thanks Hermanus for your answer.

I just want to confirm... you say i must put this commands on VoGW ?

sip-ua 
no host-registrar

I'll try and i let you know...

best regards.

had a great day . best regards, and rate if you'll find this post useful

Re: CME SIP TRUNK WITH SONUS GSX9000 HD NO REGISTER (Received :

i'll modify all sip headers to conform what the TISP requires.

voice class sip-profiles 1

request REGISTER sip-header Via modify "10.255.250.30:5060" "200.27.121.158:5060"

request REGISTER sip-header Contact modify "<>" "<56227602276>"

request REGISTER sip-header From modify "<>" "<56227602276>"

request REGISTER sip-header To modify "@(.*)>" "@200.75.6.67>"

to change from this

Sent:

REGISTER sip:signal.convergia.com:5060 SIP/2.0

Via: SIP/2.0/UDP 10.255.250.30:5060;branch=z9hG4bK251E4F

From: <099........>;tag=3351250-2248

To: <099........>

Date: Wed, 15 Jan 2014 14:57:16 GMT

Call-ID: 1DA2B6CB-7D2C11E3-8021808A-B237480

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1389797836

CSeq: 2 REGISTER

Contact: <099........>

Expires:  3600

Supported: path

Content-Length: 0

Received: 

SIP/2.0 405 Method Not Allowed

Via: SIP/2.0/UDP 10.255.250.30:5060;branch=z9hG4bK251E4F

From: <099........>;tag=3351250-2248

To: <099........>

Call-ID: 1DA2B6CB-7D2C11E3-8021808A-B237480

CSeq: 2 REGISTER

Allow: INVITE,ACK,CANCEL,BYE,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS

Content-Length: 0

to this

Sent:

REGISTER sip:200.75.6.67:5060 SIP/2.0

Via: SIP/2.0/UDP 200.27.121.158:5060;branch=z9hG4bK4B132B

From: <56227602276>;tag=3AE2AB0-17CA

To: <099........>

Date: Wed, 15 Jan 2014 16:53:19 GMT

Call-ID: 5EAE36EC-7D3C11E3-8089808A-B237480

User-Agent: Cisco-SIPGateway/IOS-12.x

Max-Forwards: 70

Timestamp: 1389804799

CSeq: 2 REGISTER

Contact: <56227602276>

Expires:  3600

Supported: path

Content-Length: 0

Received:

SIP/2.0 405 Method Not Allowed

Via: SIP/2.0/UDP 200.27.121.158:5060;branch=z9hG4bK4B132B

From: <56227602276>;tag=3AE2AB0-17CA

To: <099........>

Call-ID: 5EAE36EC-7D3C11E3-8089808A-B237480

CSeq: 2 REGISTER

Allow: INVITE,ACK,CANCEL,BYE,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS

Content-Length: 0

now the TISP accept the calling, but there is another weird behavior... the call never complettes from the CUCME side, The VoIPGW send an error

Jan 15 16:46:17.962: %VOICE_IEC-3-GW: H323: Internal Error (TCS ack wait timeout): IEC=1.1.183.5.63.0 on callID 116 GUID=59A00FCC7D3B11E398E5FC9A981BD0D7

Jan 15 16:46:17.962: //116/59A00FCC98E5/CCAPI/cc_api_call_disconnected:

   Cause Value=41, Interface=0x6966DD14, Call Id=116

Jan 15 16:46:17.962: //116/59A00FCC98E5/CCAPI/cc_api_call_disconnected:

   Call Entry(Responsed=TRUE, Cause Value=41, Retry Count=0)

Jan 15 16:46:17.962: //116/59A00FCC98E5/CCAPI/ccConferenceDestroy:

   Conference Id=0x3, Tag=0x0

Jan 15 16:46:17.966: //116/59A00FCC98E5/CCAPI/cc_api_bridge_drop_done:

   Conference Id=0x3, Source Interface=0x6966DD14, Source Call Id=116,

   Destination Call Id=117, Disposition=0x0, Tag=0x0

Jan 15 16:46:17.966: //117/59A00FCC98E5/CCAPI/cc_api_bridge_drop_done:

   Conference Id=0x3, Source Interface=0x697A9E78, Source Call Id=117,

   Destination Call Id=116, Disposition=0x0, Tag=0x0

Jan 15 16:46:17.966: //116/59A00FCC98E5/CCAPI/cc_generic_bridge_done:

   Conference Id=0x3, Source Interface=0x697A9E78, Source Call Id=117,

   Destination Call Id=116, Disposition=0x0, Tag=0x0

Jan 15 16:46:17.966: //116/59A00FCC98E5/CCAPI/ccCallDisconnect:

   Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=41)

Jan 15 16:46:17.966: //116/59A00FCC98E5/CCAPI/ccCallDisconnect:

   Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)

Jan 15 16:46:17.966: //116/59A00FCC98E5/CCAPI/cc_api_get_transfer_info:

   Transfer Number Is Null

Jan 15 16:46:17.966: //117/59A00FCC98E5/CCAPI/ccCallDisconnect:

   Cause Value=41, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)

Jan 15 16:46:17.966: //117/59A00FCC98E5/CCAPI/ccCallDisconnect:

   Cause Value=41, Call Entry(Responsed=TRUE, Cause Value=41)


anyone know why happen that ?

i currently had in the CUCME an FXO port and when i call to another number the dialpeer sents me thru that port and i had a call without any problem,

only the problem appears when i try to get communication thru the SIP trunk.

thanks for your time.

had a great day . best regards, and rate if you'll find this post useful

CME SIP TRUNK WITH SONUS GSX9000 HD NO REGISTER (Received : SIP/

can we see debug voice ccsip for this new strange behavior.

It is good news to see the ISP accepts your SIP request now.

Voice CCIE #37771

Re: CME SIP TRUNK WITH SONUS GSX9000 HD NO REGISTER (Received :

here is the debug ccsip all trace log.

i can see in log some stuff but can't understand it at all ...

had a great day . best regards, and rate if you'll find this post useful
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