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  #1 (permalink)  
Old April 30th, 2008, 09:19 AM
biknit biknit is offline
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Default Call forwarding

Hi everyone,

I'm stuck with a stupid problem concerning my asterisk server and my pstn provider. For your information, I'm moving my old telephone system to a new VoIP system. Here is my concern:

Using my old system: My pstn provider allows me to forward calls to any number using *05 prefix. Thus, when I want to forward calls to any_number, I just have to dial *05 + 9 + any_number on my phone, then I hear a confirmation tone, and it's done. All my calls are gonna be forwarded to this number. If I want to cancel this call forwarding, I just have to dial #05, then I hear a confirmation tone and it's canceled! Very simple and it works well.

Using asteriskNOW: I want to be able to use the same service with my new asteriskNOW system. In my dial plan, here is what I wrote: exten=_101XXXXXXXX,1,Dial(SIP/*059${EXTEN:3}@101)
Thus, when I dial 101 + any_number it should do the same trick as before. For instance, it should activate the call forwarding service to this any_number (8digits). But it doesn't! Instead, I've got a Receive response:"Forbidden" from '"user" <sip:75@192.168.10.233>; tag=as74e0202d'

I desperately need your help on this as I've no idea why it doesn't want to dial this *05 number!

Thanks in advance!
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Old May 2nd, 2008, 04:11 AM
biknit biknit is offline
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Default Re: Call forwarding

I thought of something yesterday. I've got a spare line in the office that I could use to forward calls. Thus, I would just have to set my SPA942 ip phone, to forward the call to any number using this line.

For example, an external number calls my extension, asterisk takes the call, send it to my phone, my phone asks asterisk to call another external number using another line and bridges the incoming to the outgoing call.

Looks simple, but when i try it, the call is forwarded properly to my mobile, it rings (but I don't hear any ring tone from the calling phone), but as soon as i pickup the call, NO MORE SOUND. I don't know why, but the sound is not transferred. I can even see in the CLI that asterisk STOPPED SOUNDS (Local/##########@context-c2e1,1 stopped sounds).. Any ideas why? And i don't understand why, it's using Local now to call, instead of SIP..

Help?

Thanks!
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