Author Topic: Sip communication with the telecommunication center  (Read 2170 times)

Offline Miss Farhad

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Sip communication with the telecommunication center
« on: November 12, 2023, 03:58:55 AM »
Hello, I have a 3300 server with version 14 that I want to establish a SIP connection between the Mitel server and the telecommunication center -
I can ping the ips from the Mitel server
SIP link is also in service
But the call does not leave or enter the server.
  The router between me and telecommunication is Mikrotik brand.
When I make an outgoing call, it shows me forbidden.
When calling the line number taken from the telecommunications company, the busy beep is played.
I received the following log during the call.
mitel server ip; 192.168.0.61
Telecom server ip: 77.104.118.110
ip microtick: 192.168.0.1
And ip 77.104.118.110 is on Nat server.
And it is tested and working on Isabelle.
                                                                               

2023-10-30  14:21:58 SSP STS->Network
OPTIONS sip:172.16.2.120:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.19:5060;branch=z9hG4bK3541072000-336160002
Max-Forwards: 70
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER,UPDATE
From: <sip:192.168.0.19:5060>;tag=0_3541072000-336160003
To: <sip:172.16.2.120;transport=udp>
Call-ID: 3541072000-336160001
CSeq: 1 OPTIONS
Contact: <sip:192.168.0.19:5060;transport=udp>
User-Agent: Mitel-3300-ICP 14.0.3.51
Content-Length: 0



2023-10-30  14:22:02 SSP STS->Network
OPTIONS sip:172.16.2.120:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.19:5060;branch=z9hG4bK3541072000-336160002
Max-Forwards: 70
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER,UPDATE
From: <sip:192.168.0.19:5060>;tag=0_3541072000-336160003
To: <sip:172.16.2.120;transport=udp>
Call-ID: 3541072000-336160001
CSeq: 1 OPTIONS
Contact: <sip:192.168.0.19:5060;transport=udp>
User-Agent: Mitel-3300-ICP 14.0.3.51
Content-Length: 0



2023-10-30  14:22:06 SSP STS->Network
OPTIONS sip:172.16.2.120:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.19:5060;branch=z9hG4bK3541072000-336160002
Max-Forwards: 70
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER,UPDATE
From: <sip:192.168.0.19:5060>;tag=0_3541072000-336160003
To: <sip:172.16.2.120;transport=udp>
Call-ID: 3541072000-336160001
CSeq: 1 OPTIONS
Contact: <sip:192.168.0.19:5060;transport=udp>
User-Agent: Mitel-3300-ICP 14.0.3.51
Content-Length: 0


2023-10-30  14:22:12 SSP App->SSP
Event Type : MakeCall
Call Object: 0 Session: 0x2eb15740 App: 2 Unique: 10000162
Realm : Local_28, User Name : system
To: sip:09354521548@77.104.118.110
From: "F.Saedeh" <sip:6143@192.168.0.19>
RequestUri: sip:09354521548@77.104.118.110:5060;transport=udp
Contact: "F.Saedeh" <sip:6143@192.168.0.19:5060;transport=udp>
Transport: udp
Asserted Identity List:
String List : "F.Saedeh" <sip:6143@192.168.0.19:5060;transport=udp>
      
 SessTimer: 90 MaxFwds: 26
Type: application/sdp
Length : 245
v=0
o=- 2173 2173 IN IP4 192.168.0.19
s=-
c=IN IP4 192.168.0.143
t=0 0
m=audio 50266 RTP/AVP 8 0 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-mitel-dtmf-type: std mitel inband



2023-10-30  14:22:12 SSP STS->Network
INVITE sip:09354521548@77.104.118.110:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.19:5060;branch=z9hG4bK3566432000-336160005
Max-Forwards: 26
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER,UPDATE
From: "F.Saedeh" <sip:6143@192.168.0.19>;tag=0_3566432000-336160006
To: <sip:09354521548@77.104.118.110>
Call-ID: 3566432000-336160004
CSeq: 1 INVITE
Min-SE: 90
Session-Expires: 90;Refresher=uas
Supported: replaces,timer
Contact: "F.Saedeh" <sip:6143@192.168.0.19:5060;transport=udp>
Content-Type: application/sdp
User-Agent: Mitel-3300-ICP 14.0.3.51
P-Asserted-Identity: "F.Saedeh" <sip:6143@192.168.0.19:5060;transport=udp>
Content-Length: 245

v=0
o=- 2173 2173 IN IP4 192.168.0.19
s=-
c=IN IP4 192.168.0.143
t=0 0
m=audio 50266 RTP/AVP 8 0 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-mitel-dtmf-type: std mitel inband


Offline lundah

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Re: Sip communication with the telecommunication center
« Reply #1 on: November 13, 2023, 12:42:41 PM »
SIP 403 - Forbidden from the carrier means just that, the endpoint is forbidden from placing the call. Verify you are sending the correct credentials, it looks like you are sending the internal username and extension, which may be the issue, but the carrier should be able to tell you what they are looking for in the INVITE message.

Online bojo387

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Re: Sip communication with the telecommunication center
« Reply #2 on: November 14, 2023, 01:55:26 AM »
Some carriers require the correct source IP (not internal) and/or the correct CLIP based in the SIP invite


 

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