IP500 v2 SIP Trunk Issue - Call Forward or Mobility

[Kbmonster]

Hello All,

I gave up on this issue and have no idea how to fix it. If anyone can help me out, I would hugely appreciate it!

Here is my issue, I have SIP trunks configured, and they are working inbound and outbound no issues. What I can’t do is forward calls or do Twinning when a call came in on SIP trunk. (I have enough licenses)

I did some necessary, testing and I can dial out up to 6 SIP lines inbound/outbound at the same time, no problems at all. After more troubleshooting, I configured the extension to dial out using the analog line for forwarding or twinning. It was working flawlessly.



720736mS SIP Call Rx: 17
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.5.102:5060;received=10.0.5.102;rport=5060;branch=z9hG4bK3af305fdd61beae093faa6d7033d16ac
From: "John B" <sip:134739834444@mkwork.onsip.com>;tag=dc3409638cdea57d
To: <sip:134739834444@mkwork.onsip.com>
Call-ID: 7500d4abe3f0572a1b7e5503728ed9ec
CSeq: 2115382568 INVITE
Server: OpenSIPS (1.10.0-notls (x86_64/linux))
Content-Length: 0

720741mS SIP Call Rx: 17
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.5.102:5060;received=10.0.5.102;rport=5060;branch=z9hG4bK3af305fdd61beae093faa6d7033d16ac
From: "John B" <sip:134739834444@mkwork.onsip.com>;tag=dc3409638cdea57d
To: <sip:134739834444@mkwork.onsip.com>;tag=9d079adf30be9ee6222f98d845a8175e.c44e
Call-ID: 7500d4abe3f0572a1b7e5503728ed9ec
CSeq: 2115382568 INVITE
Server: OpenSIPS (1.10.0-notls (x86_64/linux))
Content-Length: 0

720743mS SIP Call Tx: 17
ACK sip:134739834444@mkwork.onsip.com SIP/2.0
Via: SIP/2.0/UDP 10.0.5.102:5060;rport;branch=z9hG4bK3af305fdd61beae093faa6d7033d16ac
From: "John B" <sip:134739834444@mkwork.onsip.com>;tag=dc3409638cdea57d
To: <sip:134739834444@mkwork.onsip.com>;tag=9d079adf30be9ee6222f98d845a8175e.c44e
Call-ID: 7500d4abe3f0572a1b7e5503728ed9ec
CSeq: 2115382568 ACK
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
User-Agent: IP Office 9.1.6.0 build 153
Content-Length: 0

723448mS SIP Call Rx: 17
BYE sip:12015128228@10.0.5.102:5060;transport=udp SIP/2.0
Record-Route: <sip:199.7.173.100;lr;ftag=att82j34KSQHc;did=4dd.56e47841>
Record-Route: <sip:199.7.173.101;lr;ftag=att82j34KSQHc;did=4dd.d186f643>
Record-Route: <sip:199.7.173.199;lr;ftag=att82j34KSQHc>
Via: SIP/2.0/UDP 199.7.173.100:5060;branch=z9hG4bKd0ef.42eea2a6.0
Via: SIP/2.0/UDP 199.7.173.101:5060;branch=z9hG4bKd0ef.19ae2e4.0
Via: SIP/2.0/UDP 199.7.173.199:5060;branch=z9hG4bKd0ef.48d1a85.0
Via: SIP/2.0/UDP 199.7.173.56:5090;received=199.7.173.56;rport=5090;branch=z9hG4bK339HKKepUFgXe
Max-Forwards: 30
From: "John B" <sip:134739834444@jnctn.net>;tag=att82j34KSQHc
To: <sip:12015128228@jnctn.net>;tag=5b4a5eccbb4e796f
Call-ID: 99380d57-f744-1234-fbb3-52540018c7f7
CSeq: 96711180 BYE
Contact: <sip:gw@gw1.new-york-1.pstn.jnctn.net;gr=gw1.new-york-1.pstn>
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,UPDATE,INFO,REFER
Supported: replaces


I am Running Version 9.1.60.16.
My Current Config is broken down into parts for the same SIP Trunk 1 Incoming (11) and Outgoing (10)

Outgoing is setup like this.

Local URI - Use Credentials User Name
Contact - *
Display Name - 12015128228
PAI - *
Incoming Group - 10
Outgoing Group - 10
Max Calls per channel- 6


Thank you guys so much for taking the time.

 

[janni78]

You're not showing your INVITE which is the most interesting part of all the SIP packets.

"Trying is the first step to failure..." - Homer

 

[amriddle01]

The From and To are the same number?

 

[Kbmonster]

@ Amriddle01
Yes, Currently From and To are the same number. I was just testing with 1 cellphone at the moment. Even if i have 2 different numbers same effect things happens

@ Janni78
I will post up the INVITE later tonight have to get to my office computer.

 

[amriddle01]

It will need to be "From" a valid DDI/DID I'd wager, so you need to turn off sending original caller ID or make sure the SIP provider allows it

 

[Kbmonster]

@ Janni78 Here is the Invite below Thank you for taking your time looking at it.

@ Amriddle01 To be honest with you, i think you are totally right and from my understanding of SIP it should work the way you said it.
I am unable to figure out how to send MY Number he going out caller ID. So far I tried "1NS12015128228" and I went as far as setting up with this 1NS12015128228"@mpmillwork.onsip.com"

72789245mS SIP Call Tx: 17
INVITE sip:134739834444@mkwork.onsip.com SIP/2.0
Via: SIP/2.0/UDP 10.0.5.102:5060;rport;branch=z9hG4bK56f11ccdd2c8ceb47f8f810f4792d86c
From: "John B" <sip:134739834444@mkwork.onsip.com>;tag=5effd182991e3a6d
To: <sip:134739834444@mkwork.onsip.com>
Call-ID: f64bc06911fca508abb3b7811f35dc58
CSeq: 1993945844 INVITE
P-Asserted-Identity: "John B" <sip:134739834444@10.0.5.102:5060>
Diversion: "12015128228" <sip:Johnllbez@mkwork.onsip.com:5060>;reason=direct;screen=no;privacy=off
Contact: "12015128228" <sip:134739834444@10.0.5.102:5060;transport=udp>
Max-Forwards: 70
Proxy-Authorization: Digest username="mkwork",realm="jnctn.net",nonce="57ddff3800004dfb1bee1bb332705c91fbccdd874c921205",response="f76d2db818dd4fd131d50c4f2d080379",uri="sip:134739834444@mkwork.onsip.com",algorithm=MD5,qop=auth,
nc=00000001,cnonce="582485e76d28c1f81abc"
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY,UPDATE
Supported: timer,100rel
User-Agent: IP Office 9.1.6.0 build 153
Content-Type: application/sdp
Content-Length: 248

v=0
o=UserA 2625771869 2156468971 IN IP4 10.0.5.102
s=Session SDP
c=IN IP4 10.0.5.102
t=0 0
m=audio 49156 RTP/AVP 9 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
72789273mS SIP Call Rx: 17
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.5.102:5060;received=10.0.5.102;rport=5060;branch=z9hG4bK56f11ccdd2c8ceb47f8f810f4792d86c
From: "John B" <sip:134739834444@mkwork.onsip.com>;tag=5effd182991e3a6d
To: <sip:134739834444@mkwork.onsip.com>
Call-ID: f64bc06911fca508abb3b7811f35dc58
CSeq: 1993945844 INVITE
Server: OpenSIPS (1.10.0-notls (x86_64/linux))
Content-Length: 0

72789282mS SIP Call Rx: 17
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.5.102:5060;received=10.0.5.102;rport=5060;branch=z9hG4bK56f11ccdd2c8ceb47f8f810f4792d86c
From: "John B" <sip:134739834444@mkwork.onsip.com>;tag=5effd182991e3a6d
To: <sip:134739834444@mkwork.onsip.com>;tag=1ee758eef3cbb66f4da79e04a0e4dad7.b9e8
Call-ID: f64bc06911fca508abb3b7811f35dc58
CSeq: 1993945844 INVITE
Server: OpenSIPS (1.10.0-notls (x86_64/linux))
Content-Length: 0

 

[Kbmonster]

I wanted to thank you for your time. You guys put me on the right path! So I started monitoring logs and looking for ways to set the OUTBOUND caller ID to show my SIP line as you recommended. And it worked for Twinning. In "System/Twinning/(set my main sip trunk number) and it worked for twinning.As for call forwarding still searching for an answer

 

[janni78]

Diversion: "12015128228" <sip:Johnllbez@mkwork.onsip.com:5060>;reason=direct;screen=no;privacy=off

Usually you would have the System -> Twinning enabled, what you need to do is to set the outgoing CLI on each users SIP tab.
The shortcode should not be Ns, just N unless the SIP provider doesn't diversion header or PAI.

"Trying is the first step to failure..." - Homer