Sip trunk

[Spekkert]

Hey Guys,

I have some trouble with a sip trunk. Traffic in and out is established. Two things that arent going as i would like.

1. When i am making a call i have to dial the number and finish with a # to make the call happening. Shortcode is like this:

0N;
Dial
0N"@sipnl.net"
20

I already put the Dial delay count under system on 0, but then you have to dial very quick. I put the timer on 2 but then it doesn't do a thing.

2. When I get an incomming call (every call), the display states the following:

a= anonymous
clip info

Every call is anonymous very strange...

I hope someone can help me further with this

Greets Marcel

 

[tlpeter]

Marcel, it should be like this:

system shortcode:
0N
dail
N
line id ARS

ARS shortcode:

N
dial
N"@sip.sipnl.net"
line id trunk

this should work.
I have a trunk from them working.



Homo sapiens non urinat in ventum

honey, i fried the IP Office !!!

Sarcasm, it's only one of the services I offer.

 

[Spekkert]

Peter,

thanks voor the info. I tried this but it doenst work. When i look in SSA i see that it does not send al the digits:

16-10-10 16:02:13-836ms Netlijn = 20, Kanalen = 1, SIP-bericht = Response, Gesprek Ref = 49, Richting = Naar schakelaar, Van = 31252830011@sipnl.net, Aan = 0630@sipnl.net, Reactie = 484 Address Incomplete

As you can see it misses 6 caracters in the mobile number. Any Ideas?

gr Marcel

 

[Spekkert]

Peter,

thanks voor the info. I tried this but it doenst work. When i look in SSA i see that it does not send al the digits:

16-10-10 16:02:13-836ms Netlijn = 20, Kanalen = 1, SIP-bericht = Response, Gesprek Ref = 49, Richting = Naar schakelaar, Van = XXXXXXX@sipnl.net, Aan = 0630@sipnl.net, Reactie = 484 Address Incomplete

As you can see it misses 6 caracters in the mobile number. Any Ideas?

gr Marcel

 

[tlpeter]

Try putting in a X in front of the N

So your system shortcode is:

0N
dial
N
Line id ARS

ARS shortcode is:

XN
dial
XN"@sip.sipnl.net"
Line id sip trunk

If that does not work then create an extra shortcode like this:

06N
dial
06N"@sip.sipnl.net"
Line id sip trunk



Homo sapiens non urinat in ventum

honey, i fried the IP Office !!!

Sarcasm, it's only one of the services I offer.

 

[tlpeter]

Sorry, you need to have a ; behind the first N in the ARS.
So only the code field should have a ; after the N.
This will send all digits at once to the provider.
Sip need enblock dailing.


Homo sapiens non urinat in ventum

honey, i fried the IP Office !!!

Sarcasm, it's only one of the services I offer.

 

[Bas1234]

Try to put a sysmon trace here to see if you get RX and TX traffic.

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged
___________________________________________

 

[Spekkert]

Peter,

i got it working now, the ; did the trick.

Dit you had the same problem that i described with the anonymous caller:

2. When I get an incomming call (every call), the display states the following:

a= anonymousclip infoEvery call is anonymous very strange...

This is the only thing thats remaining for this provider that i can't get rid of...

 

[tlpeter]

I do not have that problem.

Homo sapiens non urinat in ventum

honey, i fried the IP Office !!!

Sarcasm, it's only one of the services I offer.

 

[Spekkert]

Strange,

every incomming call that I get has : contact anonymous in it.

A piece of the trace may give you an idea:

23521570mS SIP Rx: UDP 91.195.161.2:5060 -> 10.0.10.30:5060
INVITE sip:31252xxxxxx@83.98.xxx.xxx:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 91.195.161.2:5060;branch=z9hG4bK-d8754z-c489bc1f1a3f236d-1---d8754z-;rport
Via: SIP/2.0/UDP 91.195.161.2:5061;branch=z9hG4bK-2av45iqxwm6qxis3;rport=5061
Max-Forwards: 69
Record-Route: <sip:91.195.161.2;lr>
Contact: "Anonymous"<sip:91.195.161.2:5061>
To: <sip:31252xxxxxx@91.195.161.2>
From: <sip:0630xxxxxx@91.195.161.2>;tag=yfebkxs62dkqvpqm.o
Call-ID: 8AC78DD7-D88011DF-B210BB26-4900045B@91.198.151.69~1o~1o
CSeq: 464 INVITE
Expires: 300
Content-Disposition: session
Content-Type: application/sdp
User-Agent: Sippy
Content-Length: 451
cisco-GUID: 2327893226-3632271839-2167013396-2848751048
h323-conf-id: 2327893226-3632271839-2167013396-2848751048

v=0
o=Sippy 155133260 0 IN IP4 91.195.161.2
s=SIP Call
t=0 0
m=audio 51138 RTP/AVP 8 0 18 4 98 98 0 100 101
c=IN IP4 91.195.161.3
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3;annexa=yes
a=rtpmap:98 G726-32/8000
a=rtpmap:98 G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
23521575mS SIP Call Rx: 20
INVITE sip:31252xxxxxx@83.98.xxx.xxx:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 91.195.161.2:5060;branch=z9hG4bK-d8754z-c489bc1f1a3f236d-1---d8754z-;rport
Via: SIP/2.0/UDP 91.195.161.2:5061;branch=z9hG4bK-2av45iqxwm6qxis3;rport=5061
Max-Forwards: 69
Record-Route: <sip:91.195.161.2;lr>
Contact: "Anonymous"<sip:91.195.161.2:5061>
To: <sip:31252xxxxxx@91.195.161.2>
From: <sip:0630xxxxxx@91.195.161.2>;tag=yfebkxs62dkqvpqm.o
Call-ID: 8AC78DD7-D88011DF-B210BB26-4900045B@91.198.151.69~1o~1o
CSeq: 464 INVITE
Expires: 300
Content-Disposition: session
Content-Type: application/sdp
User-Agent: Sippy
Content-Length: 451
cisco-GUID: 2327893226-3632271839-2167013396-2848751048
h323-conf-id: 2327893226-3632271839-2167013396-2848751048

v=0
o=Sippy 155133260 0 IN IP4 91.195.161.2
s=SIP Call
t=0 0
m=audio 51138 RTP/AVP 8 0 18 4 98 98 0 100 101
c=IN IP4 91.195.161.3
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3;annexa=yes
a=rtpmap:98 G726-32/8000
a=rtpmap:98 G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
23521576mS Stun: Info: media ports will be mapped with STUN.
23521576mS Sip: License, Valid 1, Available 5, Consumed 0
23521578mS SIP Call Tx: 20
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 91.195.161.2:5060;branch=z9hG4bK-d8754z-c489bc1f1a3f236d-1---d8754z-;rport
Via: SIP/2.0/UDP 91.195.161.2:5061;branch=z9hG4bK-2av45iqxwm6qxis3;rport=5061
Record-Route: <sip:91.195.161.2;lr>
From: <sip:0630xxxxxx@91.195.161.2>;tag=yfebkxs62dkqvpqm.o
To: <sip:31252xxxxxx@91.195.161.2>;tag=7dc5037ad6f2689d
Call-ID: 8AC78DD7-D88011DF-B210BB26-4900045B@91.198.151.69~1o~1o
CSeq: 464 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Length: 0

23521578mS SIP Tx: UDP 10.0.10.30:5060 -> 91.195.161.2:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 91.195.161.2:5060;branch=z9hG4bK-d8754z-c489bc1f1a3f236d-1---d8754z-;rport
Via: SIP/2.0/UDP 91.195.161.2:5061;branch=z9hG4bK-2av45iqxwm6qxis3;rport=5061
Record-Route: <sip:91.195.161.2;lr>
From: <sip:0630xxxxxx@91.195.161.2>;tag=yfebkxs62dkqvpqm.o
To: <sip:31252xxxxxx@91.195.161.2>;tag=7dc5037ad6f2689d
Call-ID: 8AC78DD7-D88011DF-B210BB26-4900045B@91.198.151.69~1o~1o
CSeq: 464 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Length: 0


Do you get the same anonymous in the contact field?

Gr Marcel

 

[tlpeter]

I think this is a provider setting.


Homo sapiens non urinat in ventum

honey, i fried the IP Office !!!

Sarcasm, it's only one of the services I offer.

 

[Spekkert]

One other thing, ik keep seeing stun messages. I didn't turn stun on, but how can i turn it off?

 

[tlpeter]

Perhaps you turned it on in monitor ?
I didn't need stun to get it working.


Homo sapiens non urinat in ventum

honey, i fried the IP Office !!!

Sarcasm, it's only one of the services I offer.

 

[Spekkert]

Where can you turn stun on and off in monitor? Or do you mean show messages for stun in monitor?

In manager stun is not active, nor are there any parameters filled in on that tab...

 

[tlpeter]

In monitor go to Filter(ctrl t) and go to the tab sip.
See if there is "Stun" checked.


Homo sapiens non urinat in ventum

honey, i fried the IP Office !!!

Sarcasm, it's only one of the services I offer.