It's not syntax, I'm doing that right. It doesn't give me an error when I use AGI DEBUG, it doesn't even give a response, just goes right on to the next command. I put a "SAY NUMBER 123 #" before and after the STREAM FILE and they both work fine, returning 200 OK, etc.
[EMAIL PROTECTED] wrote:
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Today's Topics:
1. RE: How to use multiple VOIP provider trunks (Damon Estep) 2. RE: Asterisk on a dialup connection? (Kerry Garrison) 3. Re: How to use multiple VOIP provider trunks (Tim Pushor) 4. Re: Comedian Voicemail Issues (Matias G.) 5. RE: How to use multiple VOIP provider trunks (Damon Estep) 6. How to park/transfer a call received from a Queue? (Wessel de Roode) 7. pass caller ID to another application or machine. (Richard Reina) 8. RE: How to park/transfer a call received from a Queue? (Damon Estep) 9. Re: How to use multiple VOIP provider trunks (Tim Pushor) 10. Re: Asterisk on a dialup connection? (Tim Pushor) 11. RE: pass caller ID to another application or machine. (Damon Estep) 12. RE: How to use multiple VOIP provider trunks (Damon Estep) 13. Re: How to park/transfer a call received from a Queue? (Matias G.) 14. Re: pass caller ID to another application or machine. (C F) 15. RE: Asterisk on a dialup connection? (Kerry Garrison) 16. RE: small qos switch (Jim Sturtevant) 17. Re: TDM01B (Russell Handorf) 18. Re: Sipura 2000 x dual g729 channels x other choices? (Daniel Bruce Lynes) 19. Re: Sipura 2000 x dual g729 channels x other choices? ([EMAIL PROTECTED]) 20. Re: Sipura 2000 x dual g729 channels x other choices? (Andres) 21. Re: Sipura 2000 x dual g729 channels x other choices? (Andres) 22. Broadvoice getting unregistered (Courtney Couch) 23. RE: Broadvoice getting unregistered (Kerry Garrison) 24. Re: Asterisk on a dialup connection? (Saul Diaz) 25. Re: High Availability on Asterisk (Matthew Boehm) 26. Re: Broadvoice getting unregistered (Courtney Couch) 27. another voipjet question (Tim Litwiller) 28. Re: another voipjet question (Art Zemon) 29. Re: High Availability on Asterisk (Andres)
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Message: 1 Date: Sun, 27 Mar 2005 19:45:38 -0700 From: "Damon Estep" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] How to use multiple VOIP provider trunks To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii"
<snip>
I am working on a phone routing system (with duplicate/redundant routes) and I will just have a way for a user to tell the system that they want to use an alternate route for the next call.
How about the simple and traditional method,
Dial 9 for an outside line, dial 8 for an alternate outside line? Or dial nothing for an outside line, dial 9 for an alternative outside line.
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Message: 2 Date: Sun, 27 Mar 2005 18:46:02 -0800 From: "Kerry Garrison" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] Asterisk on a dialup connection? To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii"
Dialup quality is going to be very very poor to the point of not being usable most of the time. You should use a service that has a low bandwidth codec that works well like Skype or Teleo. The Codecs for Asterisk do not like dialup. I have heard that Speex might work ok but I havent tried it. Only Firefly supports it as far as I know.
Kerry http://geekgazette.com
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Sunday, March 27, 2005 6:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk on a dialup connection?
How will this fare?
I am planning on putting an asterisk box for my brother in the Philippines but they only have dialup internet. I want them to be able to use a telephone set on a phonejack or linejack card and call me and vice versa via VOIP.
My setup in the US is working already with a broadband cable connection.
I am thinking that dialup may not work because of the bandwidth required unless I can use the onbord G723.1 codecs on the quicknet cards.
Ohphone allows this through h323 I think but I want an asterisk solution. If not a fullblown asterisk install on my brothers machine, maybe set it up as a h323 client to mine.
I am currently working on setting up one of my lan machines with ohphone to connect to my asterisk box to call FWD and such. Is this possible?
Somehow asterisk must translate the codecs from whatever SIP uses to whatever ohphone uses ( I will force it to low bandwitdh g723.1).
I am hoping this will work and that the Vonage interconnect will be up soon as this will be a cheap way for them to contact my sister as well.
I am still an asterisk newbie so pardon me if the questions seem newbie-ish.
Has anybody gone down this path? I hate to have to reinvent the wheel.
Anybody have any ideas?
_______________________________________________
Might work, but not well. Just have him install a softphone on his pc, connect to the internet, and register on your * box. Use a high compression codec. It is impossible to get g.711 to work over a POTS line. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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Message: 3 Date: Sun, 27 Mar 2005 11:51:55 -0800 From: Tim Pushor <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] How to use multiple VOIP provider trunks To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Yeah something like that ;-)
Thanks, Tim
Damon Estep wrote:
<snip>
I am working on a phone routing system (with duplicate/redundant routes) and I will just have a way for a user to tell the system that they want to use an alternate route for the next call.
How about the simple and traditional method,
Dial 9 for an outside line, dial 8 for an alternate outside line? Or dial nothing for an outside line, dial 9 for an alternative outside line.
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Message: 4 Date: Sun, 27 Mar 2005 23:50:40 -0300 From: "Matias G." <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Comedian Voicemail Issues To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1"
Something similar has happened to mw once, it was just that there was some kind of a .tmp file (the one which comedian saves until you confirm your greeting msg that doesn't get renamed... I had to manually erase them and then try again and everything worked fine...
hope this helps.
bye,
M.
----- Original Message ----- From: "Carlos Lenz" <[EMAIL PROTECTED]>
To: <[email protected]>
Sent: Sunday, March 27, 2005 7:11 PM
Subject: [Asterisk-Users] Comedian Voicemail Issues
Hello, I have set up Comedian Mail on my Asterisk system. I am using Voicemail not Voicemail2 in my extensions.conf file.
The system works great except for 1 thing...It is not possible to create custom unavailable or greeting messages for 3/4 voicemail boxes.
For some odd reason 3/4 users are unable to modify the default voicemail prompt with their own custom greeting. The greeting gets recorded to the system, but for some reason the Comedian Voicemail application will not use the correct audio files.
Has any one encountered this issue? I am at a bit of a loss or how to fix it.
Are there other voicemail systems for Asterisk systems besides Comedian?
thanks, Alex
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Message: 5 Date: Sun, 27 Mar 2005 19:52:35 -0700 From: "Damon Estep" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] How to use multiple VOIP provider trunks To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii"
Not sure if you are joking or not (the smiley face confused me), you do
realize this is quite simple, right?
Yeah something like that ;-)
Thanks, Tim
Damon Estep wrote:
system that<snip>
I am working on a phone routing system (with duplicate/redundant routes) and I will just have a way for a user to tell the
they want to use an alternate route for the next call.
How about the simple and traditional method,
Dial 9 for an outside line, dial 8 for an alternate outside line? Or dial nothing for an outside line, dial 9 for an alternative outside line.
------------------------------
Message: 6 Date: Mon, 28 Mar 2005 04:55:31 +0200 From: "Wessel de Roode" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] How to park/transfer a call received from a Queue? To: <[email protected]> Cc: [EMAIL PROTECTED] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="windows-1250"
Hi!
I'm trying to transfer a incomming call from a Queue to another extension.
I'm receiving a call from a queue with the AgentCallbackLogin.
The queu is as following: Queue(sales|t)
Which should allow transfers.
So as soon as the call is answered I would like to be able to transfer it When the agent presses the # I get the dialtone but as soon as I press any digit Asterisk tells me that that is a wrong extension?
Calling between phones and park calls works fine, so the parking application is working ok. I'm only missing something here with the Queue's.
Here are my configuration fragments. extensions.conf: [incoming] include => parkedcalls exten => 1111,1,Answer exten => 1111,2,Queue(sales|t)
features.conf:
[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park calls on
context => parkedcalls
Queues.conf: [sales] joinempty = yes announce-frequency = 30 announce-holdtime = yes member => Agent/2537
Please help :-)
Thanks in advanced,
Wessel de Roode
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