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Community Member

Sip trunk to ISP ?

  

Guys I need help with connecting to my  service provider via sip. I have configured the internal lan with

Sccp and one sip dial-peer to unity module on uc500.

I have been give these credentials to use to connect to the service provider but my performance today was just a joke as I could not do it.. can someone please tell me how I should go about setting up this trunk to the provider and do I have to configure translation patters so internal users can use the single sip number when dialing out of the lan?

Here are the details :

Provider ip add: for instance, 2.2.2.2/24

Incoming phone number 2345678902

Voip password cisco

Username 2345678902

Guys any help would be appreciated. Any links are also fine please

Thanks

5 REPLIES
Community Member

Re: Sip trunk to ISP ?

Guys I managed to gather some configs.

sip-ua

registrar dns:deleted.com expires 3600
sip-server dns:deleted.com

also

dial-peer voice 1 voip

description ** Inbound Calls
voice-class sip outbound-proxy ipv4:2.2.2.2
dtmf-relay rtp-nte

I understand that outbound proxy ip address is to allow only sip connections to the telephony service provider but I do not understand the voice-class part since I did not see anything about voice class in the configuration, apart from the one shown here. I would have pasted the entire configs here but it was sent to me by a pal and it is for the place where he works.

sccp local FastEthernet0/1.10
sccp ccm 172.16.16. 1 identifier 1
sccp

guys I need to just makes sense of these commands so I can come closer to understanding. Help please guys because I may be ask to go to the customer site and install the equipment. I don’t want to be shamed. L 

Community Member

Re: Sip trunk to ISP ?

guys i still need help. here is a debug ccsip message i did . i will also upload thte running config in a sec. please can some help me look into why my calls are not coming into the lan from the Lan

Received:
INVITE sip:0201111111@2xx.xx.x.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xx.x.xxx:5060
Via: SIP/2.0/UDP xxx.xx.x.xxx:5060;branch=z9hG4bK244e6964;rport
From: "07872054320" <>078721110@xxx.xx.x.xxx>;tag=as16a09c5c
To: <>020111111@xxx.xx.x.xxx>
Contact: <>07872011111@xxx.xx.x.xxx:5060>
Call-ID: 203c5e05412084a13ec3ad3d55a53bbb@xxx.xx.7.226
CSeq: 102 INVITE
User-Agent: Zen Internet Telephony Service
Max-Forwards: 70
Date: Mon, 01 Mar 2010 19:41:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
Supported: replaces
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 4406 4406 IN IP4 212.23.x.xxx
s=session
c=IN IP4 xxx.xx.x.xxx
t=0 0
m=audio 20242 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Mar  1 19:40:19.731: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request - 'Invalid Host'
Reason: Q.850;cause=100
Date: Mon, 01 Mar 2010 19:40:19 GMT
From: "0787111111" <>0787111110@xxx.xx.x.x>;tag=as16a09c5c
Allow-Events: telephone-event
Content-Length: 0
To: <>020811111@xxx.xx.x.xxx>;tag=318EC8-1E9D
Call-ID: 203c5e05412084a13ec3ad3d55a53bbb@xxx.xx.x.xxx
Via: SIP/2.0/UDP xxx.xx.x.xxx:5060,SIP/2.0/UDP xxx.xx.x.xxx:5060;branch=z9hG4bK244e6964;rport
CSeq: 102 INVITE
Server: Cisco-SIPGateway/IOS-12.x


Mar  1 19:40:19.775: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:020811111@xxx.xx.x.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xx.xx.xxx:5060
Via: SIP/2.0/UDP xxx.xx.x.xxx:5060;branch=z9hG4bK244e6964;rport
From: "07872054320" <>0787111110@xxx.xx.x.xxx>;tag=as16a09c5c
To: <>02081xxxxxx@xxx.xx.x.xxx>;tag=318EC8-1E9D
Contact: <>0787xxxxxx@xxx.xx.x.xxx:5060>
X-ZEN-Transaction: OK
Call-ID: 203c5e05412084a13ec3ad3d55a53bbb@xxx.xx.xx.xxx
CSeq: 102 ACK
User-Agent: Zen Internet Telephony Service
Max-Forwards: 70
Content-Length: 0


Mar  1 19:40:19.783: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0208111111@xxx.xx.x.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xx.x.xxx:5060
Via: SIP/2.0/UDP xxx.xx.x.xxx:5060;branch=z9hG4bK406e3953;rport
From: "0787111111" <0787211111110>;tag=as7c6c6149
To: <>020811111111@xxxx.xx.x.xxx>
Contact: <0787111111XXX.XX.X.XXX:5060>
Call-ID: 4c41cef04e6b5ae75978f6f1495dbd18@xxx.xx.x.xxx
CSeq: 102 INVITE
User-Agent: Zen Internet Telephony Service
Max-Forwards: 70
Date: Mon, 01 Mar 2010 19:41:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
Supported: replaces
Content-Type: application/sdp
Content-Length: 287

Community Member

Re: Sip trunk to ISP ?

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
  registrar server
    outbound-proxy ipv4:xxx.xx.x.xxx (ITSP SIP PROXY ADDRESS)

dial-peer voice 11 voip
description sip-coming-in
destination-pattern 020811111
voice-class sip outbound-proxy ipv4:xxx.xx.x.xxx (ITSP SIP PROXY ADDRESS)
session protocol sipv2
session target dns:voip.zen.co.uk
session transport udp
incoming called-number 020811111
dtmf-relay rtp-nte
codec g711ulaw
no vad
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
!

dial-peer voice 12 voip
translation-profile outgoing to-Zen
destination-pattern .T
voice-class sip outbound-proxy ipv4:xxx.xx.x.xxx (ITSP SIP PROXY ADDRESS)
session protocol sipv2
session target dns:voip.zen.co.uk
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
no vad
no supplementary-service sip moved-temporarily

sip-ua
authentication username 0202222222 password 7 011E2A2C491F0B0475 realm voip.zen.co.uk
authentication username 0202222222 password 7 1104352D0506060750
no remote-party-id
registrar dns:voip.zen.co.uk expires 3600
sip-server dns:voip.zen.co.uk
host-registrar

ephone-dn  4  dual-line
number 020811111
!
!
ephone  1
device-security-mode none
mac-address 0017.0E7E.0A4E
username "max"
button  2:4

Community Member

Re: Sip trunk to ISP ?

wow sip is so tricky. i finally got a hold of it and every thing is working fine plus i have no problems with voice mail.

Hall of Fame Super Gold

Re: Sip trunk to ISP ?

Glad you resolved, sorry nobody replied but as you have seen these are issues better dealt with hands on.

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