[Asterisk-Users] AGI STREAM FILE command

Mon Mar 28 07:30:18 MST 2005

Has anyone had success with the AGI STREAM FILE command with the CVS?  I 
can't get it to work with the debian 1.0.5 package or the CVS on Redhat 
or Debian.

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.

asterisk-users-request at lists.digium.com 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?
>      (hermann at wecke.com)
>  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)
>
>
>----------------------------------------------------------------------
>
>Message: 1
>Date: Sun, 27 Mar 2005 19:45:38 -0700
>From: "Damon Estep" <damon at suburbanbroadband.net>
>Subject: RE: [Asterisk-Users] How to use multiple VOIP provider trunks
>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>	<asterisk-users at lists.digium.com>
>Message-ID:
>	<07668904BA88BA4E9DA11CDE5B594CB29EFAD1 at ns1.soho.soho-systems.com>
>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.
>
>
>
>------------------------------
>
>Message: 2
>Date: Sun, 27 Mar 2005 18:46:02 -0800
>From: "Kerry Garrison" <kerryg at techdatapros.com>
>Subject: RE: [Asterisk-Users] Asterisk on a dialup connection?
>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
>	<asterisk-users at lists.digium.com>
>Message-ID: <0MKz1m-1DFkGT1T4r-0003Sk at mrelay.perfora.net>
>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: asterisk-users-bounces at lists.digium.com
>[mailto:asterisk-users-bounces at lists.digium.com] 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.
>_______________________________________________
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>Asterisk-Users at lists.digium.com
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>
>
>
>
>------------------------------
>
>Message: 3
>Date: Sun, 27 Mar 2005 11:51:55 -0800
>From: Tim Pushor <timp at crossthread.com>
>Subject: Re: [Asterisk-Users] How to use multiple VOIP provider trunks
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>	<asterisk-users at lists.digium.com>
>Message-ID: <42470EDB.900 at crossthread.com>
>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.
>>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>To UNSUBSCRIBE or update options visit:
>>  http://lists.digium.com/mailman/listinfo/asterisk-users
>> 
>>
>>    
>>
>
>
>
>------------------------------
>
>Message: 4
>Date: Sun, 27 Mar 2005 23:50:40 -0300
>From: "Matias G." <listas_ast at reliable.com.ar>
>Subject: Re: [Asterisk-Users] Comedian Voicemail Issues
>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>	<asterisk-users at lists.digium.com>
>Message-ID: <002b01c53340$ef10db30$c900a8c0 at krikkit>
>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" <clenz at emvisiontech.com>
>To: <Asterisk-Users at lists.digium.com>
>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
>>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>    
>>
>
>
>
>------------------------------
>
>Message: 5
>Date: Sun, 27 Mar 2005 19:52:35 -0700
>From: "Damon Estep" <damon at suburbanbroadband.net>
>Subject: RE: [Asterisk-Users] How to use multiple VOIP provider trunks
>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>	<asterisk-users at lists.digium.com>
>Message-ID:
>	<07668904BA88BA4E9DA11CDE5B594CB29EFAD2 at ns1.soho.soho-systems.com>
>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:
>>
>>    
>>
>>><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.
>>>
>>>      
>>>
>
>
>------------------------------
>
>Message: 6
>Date: Mon, 28 Mar 2005 04:55:31 +0200
>From: "Wessel de Roode" <wessel at sourcelab.nl>
>Subject: [Asterisk-Users] How to park/transfer a call received from a
>	Queue?
>To: <asterisk-users at lists.digium.com>
>Cc: wessel at sourcelab.nl
>Message-ID: <200503280256.j2S2uYHJ024663 at sourcelab01.workgroup>
>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
>
>  
>