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-   -   "The number you have dialed could not be connected" (https://forum.sipbroker.com/showthread.php?t=1729)

martin 08-02-2007 11:06 AM

Quote:

Originally Posted by jzako (Post 11188)
hi,

I have disabled the cal forwarding option and the voice mail service but the problem remains.

Jani

I don't want you to disable voicemail only the call forwarding rule. I want to know if the call goes to voicemail when you turn off the call forwarding rule?
.

jzako 08-02-2007 11:28 AM

Quote:

Originally Posted by martin (Post 11189)
I don't want you to disable voicemail only the call forwarding rule. I want to know if the call goes to voicemail when you turn off the call forwarding rule?
.

I have tried that, but not goes to the voicemail....

martin 08-02-2007 12:25 PM

Quote:

Originally Posted by jzako (Post 11190)
I have tried that, but not goes to the voicemail....

If it does not go to voicemail then it is not hitting our system. Without more details it is difficult to even suggest to you what is going wrong.

IF you can provide more information maybe we can help you.
.

jzako 08-02-2007 12:38 PM

Quote:

Originally Posted by martin (Post 11193)
If it does not go to voicemail then it is not hitting our system. Without more details it is difficult to even suggest to you what is going wrong.

IF you can provide more information maybe we can help you.
.

What kind of details do you need?

martin 08-02-2007 12:44 PM

Quote:

Originally Posted by jzako (Post 11194)
What kind of details do you need?

Looking at your account you still have Voicemail off i.e. Voicemail Active*=No

You mentioned you set this to Yes. Please set this to Yes and retry your test.

Then post the details of what occurs. The more details you provide the more chance we will have to help you. Telling us "it does not work" makes it imossible for us to help you.

The key here is diagnostics and the more the better.
.

jzako 08-02-2007 05:28 PM

Quote:

Originally Posted by martin (Post 11195)
Looking at your account you still have Voicemail off i.e. Voicemail Active*=No

You mentioned you set this to Yes. Please set this to Yes and retry your test.

Then post the details of what occurs. The more details you provide the more chance we will have to help you. Telling us "it does not work" makes it imossible for us to help you.

The key here is diagnostics and the more the better.
.

I have turned on the service, but the message is the same. I have tried to call the number with some other provider without success. The message was the same. If I set up my VSP in the ATA device, it works fine. With a voxalot service, the outgoing calls are routed to the appropirate VSP according to the dial plans, but the incoming call can not be connected. What should I try or check for a successful call reception?

Jani

jzako 08-06-2007 04:05 AM

any suggestion?

ricardojpereira 08-06-2007 08:53 AM

Problem calling the DID of a registered VSP
 
Hi.
I have the same symptoms, and I had discussed the issue with the VSP already, as seen below.
Sorry for reposting this, but I believe it may be the same issue and I didn't have any answer on the other post as well.
------

Hi.
I have currently a voice provider (Netcall in Portugal) which works fine with my Sipura.
It is currently registered with Voxalot (successfully), but every time I use the associated DID it fails to hand my ATA the call.

I contacted the provider which told me that voxalot is rejecting the call with status : 484 Address Incomplete

I wonder if you could help me. Below find the two concerned messages exchanged between netcall and voxalot:

FROM 351305504628 192.168.1.20:5060
INVITE sip:351305504628@85.17.19.194 SIP/2.0
Via: SIP/2.0/UDP
~~~PROXY_SIP_IP~~~:~~~PROXY_SIP_PORT~~~;branch=z9h G4bK-gzyrq37z
Via: SIP/2.0/UDP 82.94.244.102:8006;branch=z9hG4bKm30a1v95677250
Max-Forwards: 70
CSeq: 1000 INVITE
Date: Fri, 03 Aug 2007 09:24:57 GMT
Call-ID: 2b3891mj-74x21515@82.94.244.102
Contact: <sip:351914844400@82.94.244.102:5060>
Record-Route: <sip:~~~PROXY_SIP_IP~~~:~~~PROXY_SIP_PORT~~~;lr>
Content-Type: application/sdp
Content-Length: 340
From: 00351914844400 <sip:00351914844400@netcall.pt>;tag=j372hp3p1272
To: <sip:351305504628@82.94.244.102>
X-IP: 85.17.19.194
X-PORT: 5061

v=0
o=root 6083 6083 IN IP4 82.94.244.100
s=session
c=IN IP4 195.22.26.85
t=0 0
m=audio 16060 RTP/AVP 111 18 3 8 0 101
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSuppff - - - -

[2007-08-03 10:38:41,334] INFO - Thread-39 - 01 RECEIVED FROM 82.94.244.98:5060 FROM 351305504628 85.17.19.194:5061 SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP
82.94.244.100:5060;branch=z9hG4bK-gzyrq37z;received=82.94.244.100
Via: SIP/2.0/UDP 82.94.244.102:8006;branch=z9hG4bKm30a1v95677250
From: 00351914844400 <sip:00351914844400@netcall.pt>;tag=j372hp3p1272
To: <sip:351305504628@82.94.244.102>;tag=as48771310
Call-ID: 2b3891mj-74x21515@82.94.244.102
CSeq: 1000 INVITE
User-Agent: voxaLot
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

martin 08-06-2007 09:27 AM

Quote:

Originally Posted by ricardojpereira (Post 11380)
INVITE sip:351305504628@85.17.19.194 SIP/2.0

This is exactly your problem. The VSP needs to send the INVITE to the contact URI Voxalot sends to it when it registers.

The SIP RFC states that the INVITE should look something like INVITE sip:xxxxxx@85.17.19.194:5061 SIP/2.0 where xxxxxx is your Voxalot number.
.

ricardojpereira 08-06-2007 09:39 AM

Thanks for the prompt answer.
Is it possible to supply me a snippet of the REGISTER message that Voxalot sends to the VSP (not necessarily mine), so that I come back to my VSP support ?

With the message I believe I could go back to the RFCs and point to them what they should be doing as per rfc text.

Thanks


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