Fwd: [Asterisk-Users] SIP over IAX2

Daniel Salama dsalama at user.net
Tue May 3 11:05:24 MST 2005


Sorry if this is posted twice. I sent this about an hour ago and 
haven't seen it in the list yet.

Thanks,
Daniel

Begin forwarded message:

> From: Daniel Salama <dsalama at user.net>
> Date: May 3, 2005 1:12:51 PM EDT
> To: "Tim Connolly" <tim at timsnet.com>
> Cc: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 
> <asterisk-users at lists.digium.com>
> Subject: Re: [Asterisk-Users] SIP over IAX2
>
> Ok. I've been trying to make this work to no avail.
>
> I think I have something screwed up in my original post, so I'm going 
> to try to rephrase what I'm trying to do.
>
> I have a bunch of agents connected to an * box using IAX (A1). I have 
> a separate * box (A2) that is running an IVR (AGI) script that the 
> agents need to get to. What I'm trying to do is create a extension in 
> A1 so that when dialed by the agents, they will be connected to the 
> IVR script in A2 using SIP (not IAX). I'm currently doing this using 
> IAX, but I have to implement it using SIP, for the IVR machine is 
> going to be SIP only in the near future.
>
> Here is what I have in A1:
>
> extensions.conf
> exten => 
> 1234,1,Dial(IAX2/ivr_script:ivr_script at 192.168.1.1/s at ivr_script)
> exten => 1234,2,HangUp
>
> In A2, I have:
>
> extensions.conf
> [ivr_script]
> exten => s,1,Answer
> exten => s,2,AGI(play_ivr.pl)
> exten => h,1,HangUp
>
> If I simply change IAX2 to SIP in A1, it won't work. If I replace it 
> with Dial(SIP/ivr_script at 192.168.1.1) it shows on the console:
>
> Got SIP response 404 "Not Found" back from 192.168.1.1
>
> which I could understand because there is no such SIP peer defined in 
> sip.conf.
>
> Any suggestions would be greatly appreciated.
>
> Thanks,
> Daniel
>
> On Apr 30, 2005, at 8:00 PM, Tim Connolly wrote:
>
>> Asterisk Box 2 (agents register)
>> extensions.conf
>> [agents-context]
>> exten => 1234,1,Dial(SIP/${EXTEN}@ab1)
>> exten => 1234,2,Hangup
>>
>> Asterisk Box 1
>> Sip.conf
>> [ab1]
>> type=friend
>> host=<ip of ab2>
>> context=incoming
>> canreinvite=yes
>> qualify=yes
>>
>> extension.conf
>> [incoming]
>> Exten => 1234....etc...
>>
>> -----Original Message-----
>> From: Daniel Salama [mailto:dsalama at user.net]
>> Sent: Saturday, April 30, 2005 6:50 PM
>> To: Tim Connolly
>> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [Asterisk-Users] SIP over IAX2
>>
>> I understand and I guess I know how to do that within a single box.
>>
>> If I have the following:
>>
>> Asterisk Box 1 (no agents)
>> extensions.conf
>> [test-ivr]
>> exten => s,1,AGI(play_ivr)
>> exten => s,2,Hangup
>>
>> Asterisk Box 2 (agents register)
>> extensions.conf
>> [agents-context]
>> exten => 1234,1,Dial(?????)
>> exten => 1234,2,Hangup
>>
>> Question is, when the agents dial 1234, how do I tell the application
>> to connect to the agent with context test-ivr of Asterisk_1?
>>
>> Thanks,
>> Daniel
>>
>> On Apr 30, 2005, at 7:12 PM, Tim Connolly wrote:
>>
>>> Maybe I'm missing something, but as long as you have the entension
>>> defined
>>> on the agent box to dial the extension on the IVR, you should be 
>>> okay.
>>> Just
>>> make sure the default SIP context on the IVR has that extension
>>> defined, or
>>> define the IVR box as a SIP peer.
>>>
>>>
>>>
>>> -----Original Message-----
>>> From: asterisk-users-bounces at lists.digium.com
>>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Daniel
>>> Salama
>>> Sent: Saturday, April 30, 2005 5:57 PM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: [Asterisk-Users] SIP over IAX2
>>>
>>> I have two asterisk boxes. I'm running an IVR script in one of them 
>>> and
>>> I have agents registered on the second box.
>>>
>>> I wish to create an extension on the * box where the agents are
>>> registered, so that when dialed, it will connect the agent to the IVR
>>> script on the other * box. However, I'd like for the connection to be
>>> done using SIP instead of IAX. Can anyone help me, if at all 
>>> possible,
>>> write this configuration?
>>>
>>> Thanks,
>>> Daniel
>>>
>>> _______________________________________________
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>>> Asterisk-Users at lists.digium.com
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