[asterisk-users] PSTN Channels merging with SIP channels!!!

[R_O_L_A_N_D at hotmail.com]

Hi Dave,

that actually makes sense..
I had probs in figuring out my disconnection dial tone, till the point I 
stoped trying to figure out..
so ur right that might be the problem..
thanks for your help ill give it a try :)

best,

Roland

--------------------------------------------------
From: "Dave Fullerton" <dfullertasterisk at shorelinecontainer.com>
Sent: Wednesday, November 12, 2008 8:23 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] PSTN Channels merging with SIP channels!!!

> R_O_L_A_N_D at hotmail.com wrote:
>> Hi All,
>>
>> I appreciate any help with this issue am facing.
>>
>> first of all my topology is as such:
>>
>> my asterisk box has two callcentric sip accounts on it.
>> as well as a PSTN line which is connected to asterisk through a Sipura 
>> 3102.
>>
>> now my problem is as such:
>>
>> I sometimes use my box as an international gateway.
>> that means when am home, I call my PSTN line. it directs me through an 
>> auto attendant that I've setup.
>> in it there's a "waitexten" for 8 seconds, where I enter the full 
>> international number with a preceding extension which directs this number 
>> to one of my callcentric lines.
>>
>> now this has worked ok for a while but lately am facing a major prob!
>> sometimes ill be talking to an international destination following the 
>> scenario explained above. and suddenly In the middle of my conversation 
>> both me and the other side could hear my PSTN attendant's recorded voice 
>> welcoming caller..
>> such a thing is extremely annoying to say the least and I have no idea 
>> why it's happening..
>> can anyone help out please?! any advice on how to catch what's causing 
>> this?!
>
> I had something similar happen to me using a SPA 3000 that I was using
> to bridge two PBX's together. I don't remember exactly how I resolved
> it, but the issue had something to do with the SPA not correctly
> reporting that it was busy. For example, if someone rang the PSTN port
> on the SPA it would answer and connect to an auto attendant. If asterisk
> then tried to make a call through said PSTN port rather than the SPA
> saying it was busy sometimes it would bridge the calls together and play
> the auto attendant message like you're describing. I think I ended up
> using CHANISAVAIL() in my dialplan to see if the SPA was busy rather
> than relying on the result of Dial().
>
> Hopefully that's of some help to you.
>
> -Dave
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>