[OpenSIPS-Users] External transfer fails (from Asterisk)
Peter den Hartog
peterdenhartog at gmail.com
Wed Oct 14 09:26:40 CEST 2009
Yes i did, but it makes no sence to me. I fixed it tho with giving my
opensips a private network + modem, for testing. But i have another issue
with that, i can call outside, but the inside gives to many hops, i opened a
new message about this:
http://n2.nabble.com/incoming-calls-fail-from-outside-td3821099.html#a3821099
Bogdan-Andrei Iancu wrote:
>
> Peter den Hartog wrote:
>>>>>>
>>>>>>
>>>>> Hello Bogdan,
>>>>>
>>>>> That is correct,
>>>>> in Asterisk i see nothing of a new call, or a transfer.. but the phone
>>>>> is
>>>>> creating a new call on line 2, in opensips i just see a new ongoing
>>>>> call.
>>>>> (the line 2 call) and on the outside phone i hear the asterisk
>>>>> wait/hold
>>>>> music.
>>>>>
>>>>> Is there any smart solution for this? can i just forward the complete
>>>>> call
>>>>> to opensips and let asterisk only forward it, and not create the call?
>>>>> (it
>>>>> now just does a dial to the sip member in opensips)
>>>>>
>>>>>
>>>>>
>>>> Oke a little update, i can now do blind (cold) transfers from asterisk
>>>> to
>>>> opensips (outside lines) but not hot transfers, then the call gets
>>>> disconnected.
>>>>
>>>>
>>> Do you see some NOTIFY requests going around? they are used during
>>> attended transfer to inform on the new call state.....
>>>
>>>
>>>
>> Nope, no NOTIFY requests.
>>
>> Well wat is ment was making asterisk dumb, and just let if forward a
>> complete call.. so instead of doing a dial to an opensips extention, just
>> make a full transfer of the call to the opensips server, and then to the
>> extention.
>>
>> I'm trying it the other way arround now, as you said earlier that the
>> opensips recieves all the calls (so is directly connected to the sip
>> trunk)
>> but i have some strange issue's with that 2, i can't call outside and
>> when i
>> call inside, the phone rings (i just made a alias) and then i can't pick
>> it
>> up or anything, the phone doesn't respond!
>>
> Hi Peter,
>
> have you checked the SIP trace to see why the call is not established
> when you pick up the ringing phone ?
>
> Regards,
> Bogdan
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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