Re: SIP trunk register

cm6043 · ‎10-07-2009

Hi

I have make sip to a service provider in USA but I can't succeed in register. Please refer to the following debug ccsip message.

----------------------------------------

001331: Oct 8 10:40:58.319 HK: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

REGISTER sip:63.129.63.yyy:5060 SIP/2.0

Via: SIP/2.0/UDP 220.184.24.xxx:5060;branch=z9hG4bKA312CC

From: <sip:9*@63.129.63.yyy>;tag=4E8B9C0-2393

To: <sip:9*@63.129.63.yyy>

Date: Thu, 08 Oct 2009 02:40:58 GMT

Call-ID: E95749D4-B2E211DE-807AA4D2-2EF52498

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

Max-Forwards: 70

Timestamp: 1254969658

CSeq: 19 REGISTER

Contact: <sip:9*@220.184.24.xxx:5060>

Expires: 3600

Content-Length: 0

-------------------------------------

001476: Oct 8 10:51:00.715 HK: //1327/000000000000/SIP/Error/act_sent_register_wait_100: act_sent_register_wa

it_100: Out of retries

001477: Oct 8 10:51:00.715 HK: //1327/000000000000/SIP/Error/ccsip_api_register_result_ind: Message Code Clas

s 4xx Method Code 100 received for REGISTER

-----------------------------

Thanks

Leung Che Man

virverma · ‎10-07-2009

Could you explain this more,

how you going to make SIP trunk to SP,

What are the parameters involved in making a SIP trunk from customer directly to SP and how SP is allowing it?

cm6043 · ‎10-07-2009

Hi Virverma,

I have the username / password and SIP server IP. Please find my config of SIP trunk for your reference.

---------------------------------

interface FastEthernet0/0

ip address 220.184.24.xxx 255.255.255.240

ip virtual-reassembly

speed 100

full-duplex

max-reserved-bandwidth 100

voice service voip

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 refer

fax protocol t38 nse force ls-redundancy 0 hs-redundancy 0 fallback cisco

h323

sip

!

voice class codec 729

codec preference 1 g711ulaw

codec preference 2 g729br8

codec preference 3 g711alaw

!

voice class codec 711

codec preference 1 g711ulaw

codec preference 2 g729br8

codec preference 3 g711alaw

!

dial-peer voice 888888 voip

description **Outgoing Call to SIP Trunk**

huntstop

preference 2

service session

destination-pattern 7T

progress_ind setup enable 3

no modem passthrough

voice-class codec 729

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

fax protocol t38 nse force ls-redundancy 0 hs-redundancy 0 fallback cisco

no vad

!

gateway

timer receive-rtp 1200

!

sip-ua

hookflash-info

authentication username 2024491446 password 014156560F5F5F5E751818

no remote-party-id

set pstn-cause 47 sip-status 486

retry invite 2

retry response 3

retry bye 3

retry prack 6

retry options 0

timers expires 300000

mwi-server ipv4:63.129.63.yyy expires 3600 port 5060 transport udp unsolicited

registrar ipv4:63.129.63.yyy expires 3600

sip-server ipv4:63.129.63.yyy

!

--------------------------------

Thanks

Leung Che Man

KonradStepniewski · ‎10-08-2009

Try to use something simple as below, then check show sip reg st

sip-ua

credentials username x password y realm IP

authentication username x password y

registrar ipv4:IP expires 3600

sip-server ipv4:IP

steven.lass · ‎10-08-2009

The SP likely wants the "From" header to contain the DID they gave you. You are currently sending them "9*" in the From header. Try changing the POTS dial peer to the SP DID to see if that will put the DID in the From header and trigger the SP to send a 401 or 407 response.

cm6043 · ‎10-08-2009

Hi Steven,

I have made some change in the config but the register is still failed. Could you let me know whether the username in dial peer 8 and sip ua should be same ? If not, what value I want to put. SP just provde me the 2024491446 and username/ password.

-----------------------------

dial-peer voice 8 pots

destination-pattern 2024491446

port 0/0/0

authentication username 2024491446 password 124AAAAAAAAA72

!

sip-ua

credentials username 2024491446 password 15405B5EABC%%57C6763 realm 63.129.63.yyy

authentication username 2024491446 password 014$%#@CC0F5F5F5E751818

--------------------------------

Line peer expires(sec) registered

============ ============= ============ ===========

2024491446 8 154 no

Thanks

Leung Che Man

steven.lass · ‎10-09-2009

What does "debug ccsip messages" show now? Is the From header user portion 2024491446? Is the SP sending back any response?

The usual suspects are:

1) firewall/NAT - I assume all is good.

2) TCP vs. UDP - I think most/all SP support UDP.

3) Message config - This is the From header.

4) Bad SP info - Did the SP give you the right IP address? You could use X-lite to verify that the SP information is correct. The SP may have instructions on how to setup X-lite or some other free client.

cm6043 · ‎10-11-2009

Hi,

I changed the sip server IP. I can make a incoming call from 1441 to 1446, it is successful. but outgoing call is failed.

---------------------------

013041: Oct 12 11:21:33.254 HK: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

REGISTER sip:xx.xx.xx.xx:5060 SIP/2.0

Via: SIP/2.0/UDP 220.yy.yy.yy:5060;branch=z9hG4bK4F8D17

From: <>2024491446@xx.xx.xx.xx>;tag=E7393D4-DC9

To: <>2024491446@xx.xx.xx.xx>

Date: Mon, 12 Oct 2009 03:21:33 GMT

Call-ID: 371BCEA4-B60F11DE-81069715-5CDDDD4A

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

Max-Forwards: 70

Timestamp: 1255317693

CSeq: 17 REGISTER

Contact: <>2024491446@220.yy.yy.yy:5060>

Expires: 3600

Content-Length: 0

013042: Oct 12 11:21:33.506 HK: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 404 Not found

From: <>2024491446@xx.xx.xx.xx>;tag=E7393D4-DC9

To: <>2024491446@xx.xx.xx.xx>;tag=1182030281-1255317693379

Call-ID: 371BCEA4-B60F11DE-81069715-5CDDDD4A

CSeq: 17 REGISTER

Via: SIP/2.0/UDP 220.yy.yy.yy:5060;branch=z9hG4bK4F8D17

Content-Length: 0

--------------------------

Thanks

Leung Che Man

cm6043 · ‎10-12-2009

Hi

I find that the xx.xx.xx.xx of from header is the service provider ip not our ip. How can I change this IP ?

From: <>2024491446@xx.xx.xx.xx>;tag=E7393D4-DC9

Thanks a lot

Leung Che Man

steven.lass · ‎10-12-2009

Try adding to your voip dialpeer:

dial-peer voice 888888 voip

voice-class sip localhost dns:serviceprovider.com

cm6043 · ‎10-12-2009

Hi Steven,

The localhost added but the From header is still not changed. Is it the reason (From header not my source IP) I can't register to SIP ?

------------------------------------------

000149: Oct 13 09:36:22.554 HK: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

REGISTER sip:xx.xx.xx.xx:5060 SIP/2.0

Via: SIP/2.0/UDP aa.aa.aa.aa:5060;branch=z9hG4bK1448A

From: <>2024491446@xx.xx.xx.xx>;tag=16CF98-100D

To: <>2024491446@xx.xx.xx.xx>

000152: Oct 13 09:39:23.029 HK: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 404 Not found

------------------------------------

Thanks

Leung Che Man

steven.lass · ‎10-13-2009

Yes, many of the SP want/expect the From header to be DID@serviceprovider.com. You would have to ask the SP for the definitive answer.

Ooops, the REGISTER message does not use that dial-peer, so you will have to make the setting global. Try:

voice service voip

sip

localhost dns:serviceprovider.com