[Asterisk-Users] Connecting 2 Asterisk using SIP

[xcel]

I did use IAX2 but sound quality wasn't that good which codec are you using with IAX2 ?


*********** REPLY SEPARATOR ***********

On 12/6/2005 at 9:22 PM Alvaro Parres wrote:
Why using SIP instead of IAX2 ???

Only a question becouse i always use IAX



 
On 12/6/05, Waldo Rubinstein <waldo at trianet..net> wrote: 
Well... not so perfectly.

What I'm experiencing is that during certain call volumes, many calls
go thru from box1 to box2. However, there are some cases where I get
this message:

Dec  6 11:11:19 WARNING[203]: chan_sip.c:9525 handle_response_invite:
Forbidden - wrong password on authentication for INVITE to 
'"5095551212" <sip:5095551212 at 10.0.0.1>;tag=as3e387d65'

and the caller gets busy signal. However, other callers calling the
same number get thru with no problems. Why is this? 

Thanks,
Waldo

On Dec 5, 2005, at 10:30 AM, Waldo Rubinstein wrote:

> This worked perfectly.
>
> Thanks,
> Waldo
>
> On Dec 5, 2005, at 4:32 AM, xcel wrote:
> 
>>
>> Try this
>>
>> ___________________________________
>> 1st Machine sip.conf
>>
>> [box2]
>> username=box1
>> type=friend
>> host= 10.0.0.2
>> secret=*****
>>
>> in extensions.conf
>>
>> exten => _XXXXXX,1,Dial(SIP/box2/${EXTEN})
>>
>> __________________________________ 
>> 2nd Machine sip.conf
>>
>> [box1]
>> username=box2
>> type=friend
>> host=10.0.0.1
>> secret=*****
>>
>> in extensions.conf
>> exten => _XXXXX,1,Dial(SIP/box1/${EXTEN})
>>
>> --xce
>>
>>
>> *********** REPLY SEPARATOR  ***********
>>
>> On 12/5/2005 at 12:11 AM Waldo Rubinstein wrote: 
>>
>>> I have 2 Asterisk servers running 1.2.0. One of them is a PSTN
>>> gateway. Currently they are connected using IAX2. I wanted to play
>>> with SIP.
>>>
>>> I setup a sip entry (type=friend) in the PSTN gateway box and a sip 
>>> entry (type=user) in the second box in order to send calls using SIP
>>> to the second box. This works fine. However, when I setup the second
>>> box as type=friend in order for it to be able to send calls back to 
>>> the gateway box, then calls no longer work from gateway box to the
>>> second box. The reported error is:
>>>
>>> Dec  5 00:07:14 NOTICE[203]: chan_sip.c:9514 handle_response_invite: 
>>> Failed to authenticate on INVITE to '"2125551212" <sip:
>>> 2125551212 at 10.0.10.1>;tag=as0698b1b9'
>>>
>>> In the gateway box, my sip.conf looks like this:
>>>
>>> [general]
>>> allowguest=yes
>>> autocreatepeer=no
>>>
>>> [secondbox]
>>> type=friend
>>> host= 10.0.0.2
>>> secret=mysecret
>>>
>>> In the second box, my sip.conf looks like this:
>>>
>>> [general]
>>> allowguest=yes 
>>> autocreatepeer=no
>>>
>>> [secondbox]
>>> type=user
>>> host=10.0.0.1
>>> secret=mysecret
>>>
>>> Any ideas on how to correctly set this up? 
>>>
>>> Thanks,
>>> Waldo
>>> _______________________________________________
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>>
>>
>>
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