Re: [asterisk-users] Open Asterisk Exchange Project
Michael, Could you elaborate ? Regards 2007/12/7, Michael Munger [EMAIL PROTECTED]: Is there anyone interested in developing an open source Asterisk / MS Exchange solution? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] Attachment encrypted? click herehttp://www.highpoweredhelp.com/tutorials/wincrypt/ . ___ --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 ___ --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
Re: [asterisk-users] Asterisk SIP Microsoft Outlook Integration
Snap latest version dates is 14 months old. Unfortunatly, it poorly supports localization (in fact, none is provided) and it's unusable for screen popup, due to lack of flexibility in regexp parsing. And it's rather slow. Beside that, it has an interesting features range. Cheers ___ --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
Re: [asterisk-users] Pickup cmd
Hi,
Your extension 100 doesn't exist in the context where you have your PickUp
instruction.
You must include the context containing your phones into the context used by
your PickUp instruction or the reverse, or precise the context to use with
PickUp by adding it into the instruction line :
[BLF_group_pickup]
exten = _**1XX,1,Pickup(${EXTEN:[EMAIL PROTECTED])
exten = _**1XX,n,Hangup
Best Regards,
Francois BERGERET
France
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] la part de Rilawich Ango
Envoyé : vendredi 7 décembre 2007 10:49
A : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : [asterisk-users] Pickup cmd
Hi all,
I have a GXP2000 with BLF configured. I follow the configuration
guide to enable the pickup cmd as follow and include it under
corresponding content.
[BLF_group_pickup]
exten = _**1XX,1,Pickup(${EXTEN:2})
exten = _**1XX,n,Hangup
The I press the single key to pickup the call to extension 100 when
there is a call to it. From CLI, I can see it issue **100 to asterisk
but failed to pickup the call.
-- Executing [EMAIL PROTECTED]:1] Pickup(SIP/102-08373480, 100)
in new stack
[Dec 7 16:47:42] NOTICE[31079]: app_directed_pickup.c:159
pickup_exec: No target channel found for 100.
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/102-08373480, ) in new
stack
Anyone can tell me if I make something wrong for the pickup cmd?
asterisk version: 1.4.15
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[asterisk-users] Installing/configuring TE120P debian way
Hi all I use asterisk (1.2 brach) from debian official packages and it works fine. Now I need to install and configure a Digium TE120P card, but I cannot find any guide to install it using debian packages. I would like to know if anyone of you knows about packages that would include the necessary kernel modules or any other method that won't be broken when the asterisk packages are updated. Would anyone consider just install everything from source (branch 1.4) as the best option? I would like to keep an easy upgradeable system like Debian packages, but could use source code if necessary. Cheers, -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ --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
Re: [asterisk-users] Installing/configuring TE120P debian way
On Sun, Dec 09, 2007 at 03:53:05PM +, Andres Jimenez wrote: Hi all I use asterisk (1.2 brach) from debian official packages and it works fine. Now I need to install and configure a Digium TE120P card, but I cannot find any guide to install it using debian packages. AFAIK those cards are not supported in the version of Zaptel in Etch. So you basically need a newer zaptel . Last time I tried the same zaptel 1.4 worked with recent asterisk 1.2, but I'm not sure about asterisk 1.2.13 . I would like to know if anyone of you knows about packages that would include the necessary kernel modules or any other method that won't be broken when the asterisk packages are updated. Etch backports of both zaptel *and* asterisk are available at http://pkg-voip.buildserver.net/ or from http://updates.xorcom.com/rapid . The latter updates less frequently and is far less complete. But includes pre-built zaptel-modules packages for etch kernels (to save you 'm-a a-i zaptel'). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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
Re: [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.
Try to implement '#include' and '#exec' in a sane way with XML. You can't just include one valid XML in another. You have to make a partial XML. And apitting it out is usually way more complicated. Furthermore, there is the issue of partial processing: do you opt for one big XML file? Or continue with one XML file per .conf file? I'm pretty sure you can include one valid XML entity in another. This functionality exists in SGML as well. I've seen it done both with parser support and also simply defining your own entity; (you can define an entity in place or in another file somewhere else), which is relatively easily, and then referring to that entity elsewhere in your document. I found a nice IBM reference to using entities at http://www.ibm.com/developerworks/xml/library/x-tipgentity.html. In fact, I'd argue XML includes are more like the dialplan's idea of inclusion when compared to includes in something like GCC. :) Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 ___ --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
Re: [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.
On Sun, Dec 09, 2007 at 01:48:58PM -0500, Martin Smith wrote: Try to implement '#include' and '#exec' in a sane way with XML. You can't just include one valid XML in another. You have to make a partial XML. And apitting it out is usually way more complicated. Furthermore, there is the issue of partial processing: do you opt for one big XML file? Or continue with one XML file per .conf file? I'm pretty sure you can include one valid XML entity in another. This functionality exists in SGML as well. I've seen it done both with parser support and also simply defining your own entity; (you can define an entity in place or in another file somewhere else), which is relatively easily, and then referring to that entity elsewhere in your document. I found a nice IBM reference to using entities at http://www.ibm.com/developerworks/xml/library/x-tipgentity.html. http://jabberd.jabberstudio.org/2/docs/jabberd_guide.html#4_4 How long does it take to spot the broken XML tagging there? (for a human eye)? How many editor steps does it take to add the extra parameter foobar? What happens if you accidentally terminate it with /fooba? How many editing steps does it take to add a comment? What happens if you delete to the end of the line end kill the end of the comment? That scary config file is one of the reasons why I dislike jabberd. XML is just too easy to get wrong. When you're parsing it - it's cool. When you're writing it: it's not. XML is just overly verbose. In fact, I'd argue XML includes are more like the dialplan's idea of inclusion when compared to includes in something like GCC. So? I don't need dialplan include (which has its own runitme overhead). I need a dumb and simple include. Not some over-smart thing that will not let me include two sections at once etc. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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
Re: [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.
What would you be reinventing? Asterisk can already get its configuration from a MySQL database. You could even add extra fields in that case to store the phone model and macaddress and integrate that into your own provisioning tools. Your application would then retrieve the configuration from the Asterisk SIP database, parse it and generate configurations for your phones. Very straight forward. The other option is create your own database with your own schema and then design your parser to create the asterisk configuration files and phone configuration files. This method has the advantage of not requiring any changes to your asterisk configuration. On Dec 7, 2007 9:12 PM, Philip Prindeville [EMAIL PROTECTED] wrote: That's sort of my point: that you have to reinvent it, and it's easy to get wrong. ___ --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
Re: [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.
Please understand that I was not making an argument for using XML configuration files. I was only pointing out that includes are possible. I'm just trying to keep the discussion to the facts we know :). I'd argue that if you're editing XML by hand (in the console or a GUI), you're doing something wrong. XML is to data, what Java is to program code, for me. I'm sure you can find many XML editors (console and GUI) that will never show you brackets or comments in the format you're complaining about. I wouldn't write Java source in nano either... but that doesn't mean JVM bytecode sucks. Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Sunday, December 09, 2007 2:21 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Using XML for configuration management,single-source-of-truth, etc. On Sun, Dec 09, 2007 at 01:48:58PM -0500, Martin Smith wrote: Try to implement '#include' and '#exec' in a sane way with XML. You can't just include one valid XML in another. You have to make a partial XML. And apitting it out is usually way more complicated. Furthermore, there is the issue of partial processing: do you opt for one big XML file? Or continue with one XML file per .conf file? I'm pretty sure you can include one valid XML entity in another. This functionality exists in SGML as well. I've seen it done both with parser support and also simply defining your own entity; (you can define an entity in place or in another file somewhere else), which is relatively easily, and then referring to that entity elsewhere in your document. I found a nice IBM reference to using entities at http://www.ibm.com/developerworks/xml/library/x-tipgentity.html. http://jabberd.jabberstudio.org/2/docs/jabberd_guide.html#4_4 How long does it take to spot the broken XML tagging there? (for a human eye)? How many editor steps does it take to add the extra parameter foobar? What happens if you accidentally terminate it with /fooba? How many editing steps does it take to add a comment? What happens if you delete to the end of the line end kill the end of the comment? That scary config file is one of the reasons why I dislike jabberd. XML is just too easy to get wrong. When you're parsing it - it's cool. When you're writing it: it's not. XML is just overly verbose. In fact, I'd argue XML includes are more like the dialplan's idea of inclusion when compared to includes in something like GCC. So? I don't need dialplan include (which has its own runitme overhead). I need a dumb and simple include. Not some over-smart thing that will not let me include two sections at once etc. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 ___ --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
Re: [asterisk-users] Open Asterisk Exchange Project
On Dec 7, 2007 11:28 AM, Michael Munger [EMAIL PROTECTED] wrote: Is there anyone interested in developing an open source Asterisk / MS Exchange solution? Please explain. This sounds interesting. But why MS exchange only? I think it's safe to say with good IMAP and LDAP support we can integrate with just about any decent enterprise messaging system. Think of all the Scalixes and Zimbras of the world. ___ --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
Re: [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.
On Dec 9, 2007 2:05 AM, Philip Prindeville [EMAIL PROTECTED] wrote: ..snip.. You think that an Asterisk configuration is a lot larger than a Cisco 5850 Access Server or a 7216 core router? IOS doesn't use XML for configuration. What's a 7216? -- Kristian Kielhofner ___ --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
Re: [asterisk-users] Open Asterisk Exchange Project
Well, we can already integrate to major platforms via SMTP. The real value is in deep integration to the most popular email platform in business: Exchange. I would love to see smart Exchange integration, where deleting the VM attached email will delete the corresponding message from asterisk. My clients would eat that up. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, December 09, 2007 2:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Open Asterisk Exchange Project On Dec 7, 2007 11:28 AM, Michael Munger [EMAIL PROTECTED] wrote: Is there anyone interested in developing an open source Asterisk / MS Exchange solution? Please explain. This sounds interesting. But why MS exchange only? I think it's safe to say with good IMAP and LDAP support we can integrate with just about any decent enterprise messaging system. Think of all the Scalixes and Zimbras of the world. ___ --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 ___ --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
[asterisk-users] Don't enter a queue if no one is logged in
I currently have the following setup: exten = 2000,1,Playback(/var/lib/asterisk/sounds/Greeting) exten = 2000,2,Queue(Qabcdef|t) exten = 2000,3,Playback(/var/lib/asterisk/sounds/EveryonesBusy) exten = 2000,4,Hangup exten = 2000,103,Hangup What happens is, that the greeting in step one is played regardless if anyone is logged into the queue. So immediately after the greet, we tell them we can't help them. What I would like is to check first to see if there is anyone logged into the queue, and then play the greeting. Is this possible? Is there a function that checks if anyone is logged in? ___ --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
Re: [asterisk-users] Open Asterisk Exchange Project
On Sun, Dec 09, 2007 at 03:30:13PM -0500, Michelle Dupuis wrote: Well, we can already integrate to major platforms via SMTP. The real value is in deep integration to the most popular email platform in business: Exchange. I would love to see smart Exchange integration, where deleting the VM attached email will delete the corresponding message from asterisk. My clients would eat that up. IMAP support for voicemail? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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
Re: [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.
Neat. Does that reference mention anything about how XML was originally designed as a generic storage mechanism for data being moved from one architecture to another where the original meaning of the data might be lost due to differences of those architectures? I suppose a case can be made for using XML to transfer a SIP phone config from the tftp server to the phone itself but it makes no sense what-so-ever (to me at least) to use XML for a static config sitting on a server. It comes across to me about the same as someone saying they think using ftp would be a neat way to transfer rtp in a SIP conversation. It really doesn't matter if it can be done or not; why bother?? On Dec 9, 2007, at 1:48 PM, Martin Smith wrote: Try to implement '#include' and '#exec' in a sane way with XML. You can't just include one valid XML in another. You have to make a partial XML. And apitting it out is usually way more complicated. Furthermore, there is the issue of partial processing: do you opt for one big XML file? Or continue with one XML file per .conf file? I'm pretty sure you can include one valid XML entity in another. This functionality exists in SGML as well. I've seen it done both with parser support and also simply defining your own entity; (you can define an entity in place or in another file somewhere else), which is relatively easily, and then referring to that entity elsewhere in your document. I found a nice IBM reference to using entities at http://www.ibm.com/developerworks/xml/library/x-tipgentity.html. In fact, I'd argue XML includes are more like the dialplan's idea of inclusion when compared to includes in something like GCC. :) Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 ___ --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 ___ --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
[asterisk-users] [DB] Using SQLite instead of AST?
Hello The DB() application is fine as long as we don't need more than one value pointed to by a key, ie. the way SleepyCat works. Problem is, for each phone number, I'd like to map more than one column, eg. name, e-mail, fax, etc. Is there a way to have DB() use SQLite instead of AST, or a way to keep both in sync (ie. export data from AST to SQLite automatically)? Thank you. ___ --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
Re: [asterisk-users] Function vs. Application?
On Fri, 7 Dec 2007 15:12:03 -0600, Tilghman Lesher [EMAIL PROTECTED] wrote: You could also think of it as the difference between a procedure and a function. [...] Unlike other languages, in Asterisk, the return value of a function may not be directly ignored (i.e. you HAVE to get it, even if you do nothing with it). Thanks for the clarification. ___ --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
Re: [asterisk-users] Asterisk SIP Microsoft Outlook Integration
OutCall is very confusing. The user has multiple options Call Contact the outlook option remains present (and does not work). That is confusing. You need a TAPI Driver for the easiest user experience. This plugs in to the Windows/Outlook framework. Not only does it work with the factory Outlook options any application that uses TAPI for placing calls will work with no added configuration/modification. There are various 3rd party TAPI drivers but I think these little things are items that need to be added to the Asterisk development. On Dec 5, 2007 12:26 PM, Jared Smith [EMAIL PROTECTED] wrote: On Wed, 2007-12-05 at 11:45 -0500, Michael Melia Jr. wrote: Does anyone know how I could integrate my Asterisk setup with Outlook One of the more popular ones seems to be Outcall, which is now open-source and available from http://outcall.sourceforge.net. I haven't tried it personally, so your mileage may vary. -- Jared Smith Community Relations Manager Digium, Inc. ___ --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 ___ --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
Re: [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.
Kristian Kielhofner wrote: On Dec 9, 2007 2:05 AM, Philip Prindeville [EMAIL PROTECTED] wrote: ..snip.. You think that an Asterisk configuration is a lot larger than a Cisco 5850 Access Server or a 7216 core router? IOS doesn't use XML for configuration. What's a 7216? Actually, it does I just don't know if that ever got exposed to the public or not. (Of course, the customers that wanted XML-based configs also wanted ION, so it might only have been exposed on ION.) While I was there (2000-2005) there was a big effort to have all config files be represented as XML. If you ever tried to diff two configs of a 7216 (VXR/CMTS) that were from different releases, you'd know why. This would drive customers crazy: portions of config would move around, whitespace would appear where it wasn't previously, names of commands would gratuitously change, etc. Parse-trees would be easier to diff. There were some interesting efforts floating around to have the configuration be sucked up via XML/secure-RPC to a client, edited there, and then pushed back into the router/IAD/firewall, what-have-you. The IOS parser was one of the hairiest pieces of code I had ever had to maintain. But we're getting off the subject. -Philip P.S. I know zip about Skinny, so don't ask... ___ --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
[asterisk-users] Dual-home Wifi/GSM phones for North America
So, for the hotel project, what was the conclusion? I don't remember seeing a summary. And were any of the phones combination Wifi/GSM? Like a Nokia E70 or E61i? I've been looking for such a phone to use myself, but so far haven't found one that I liked. Common flaws were: * poor standby time * lousy interface for doing complex configurations (esp. certificate management) * poor external provisioning documentation or requiring proprietary (and Windows-based) tools * inadequate SIP implementation * inability to hand-off between Wifi and GSM (or reverse) seamlessly Has anyone found a product that overcomes these issues? And what's the word on a Wifi/SIP client for the iPhone? -Philip ___ --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
Re: [asterisk-users] Open Asterisk Exchange Project
Tzafrir wrote: On Sun, Dec 09, 2007 at 03:30:13PM -0500, Michelle Dupuis wrote: Well, we can already integrate to major platforms via SMTP. The real value is in deep integration to the most popular email platform in business: Exchange. I would love to see smart Exchange integration, where deleting the VM attached email will delete the corresponding message from asterisk. My clients would eat that up. IMAP support for voicemail? That would be an option if Microsoft had not implimeted such a limited set of IMAP features in Exchange. Exchange does not support the concept of master-user when using IMAP, so you'd need to have the password of every Exchange account you wanted to integrate with Asterisk voicemail. I suppose that the addition of IMAP support to Asterisk's voicemail application might open the way for someone to look at using a MAPI library, such as the one being developed by the OpenChange folks. Dan ___ --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
Re: [asterisk-users] [DB] Using SQLite instead of AST?
On Sunday 09 December 2007 15:18:13 Vincent wrote: The DB() application is fine as long as we don't need more than one value pointed to by a key, ie. the way SleepyCat works. Problem is, for each phone number, I'd like to map more than one column, eg. name, e-mail, fax, etc. Is there a way to have DB() use SQLite instead of AST, or a way to keep both in sync (ie. export data from AST to SQLite automatically)? No, but you can use func_odbc with a backend SQLite driver. -- Tilghman ___ --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
[asterisk-users] One server, multiple companies
Hello all,
Just starting to setup asterisk v 1.4.11 and need to run three distinct phone
systems for three different companies.
So far, I have inbound lines going to the appropriate dial plan within the
extensions.conf file. I'm using
exten = _X.,1,NoOp(FROM NUMBER: ${SIP_HEADER(TO):5:10})
to determine which number is being dialed by the caller and then using a gotoif
to get to correct greeting (correct company).
My question is... lets assume all three companies have extension numbers being
2000, 2001 2002, how does one separate them?
Or, lets say the extensions are:
company A -- 2000, 2001,2002
company B -- 3000, 3001, 3002
company C -- 4000, 4001, 4002
Since they're on one server with one asterisk process, how can I use context
correctly so that the user at 4002 cannot get through to the user at company A
whose extension is 2000 as currently, I can dial 2000 from phone 4002.
That's my current problem, how should this be setup? Is my architecture
correct? Should I be running different processes for each company? Can context
resolve what I need?
Please advise.
thanks,
Otto
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Re: [asterisk-users] One server, multiple companies
Eric C. wrote:
Hello all,
Just starting to setup asterisk v 1.4.11 and need to run three distinct phone
systems for three different companies.
So far, I have inbound lines going to the appropriate dial plan within the
extensions.conf file. I'm using
exten = _X.,1,NoOp(FROM NUMBER: ${SIP_HEADER(TO):5:10})
to determine which number is being dialed by the caller and then using a
gotoif to get to correct greeting (correct company).
My question is... lets assume all three companies have extension numbers
being 2000, 2001 2002, how does one separate them?
Or, lets say the extensions are:
company A -- 2000, 2001,2002
company B -- 3000, 3001, 3002
company C -- 4000, 4001, 4002
Since they're on one server with one asterisk process, how can I use context
correctly so that the user at 4002 cannot get through to the user at company
A whose extension is 2000 as currently, I can dial 2000 from phone 4002.
That's my current problem, how should this be setup? Is my architecture
correct? Should I be running different processes for each company? Can
context resolve what I need?
Please advise.
thanks,
Otto
First off, *nuke* the default context in sip.conf, extensions.conf, and
voicemail.conf ... it will just get you into trouble!
I do something like in my extensions.conf file:
[incoming]
exten = 208229,1,Goto(s,1,incoming-acme)
exten = 208229,1,Goto(s,1,incoming-fido)
exten = 208229,1,Goto(s,1,incoming-big-jims)
...
[incoming-acme]
exten = s,1,Answer()
exten = s,n,Wait(0.75)
exten = s,n(greeting),Playback(brief-directory-acme)
exten = s,n(exten),Background(vm-enter-num-to-call)
exten = s,n,WaitExten(5)
exten = s,n(goodbye),Playback(vm-goodbye)
exten = s,n(end),Hangup()
; these are the extensions that are exposed both to internal callers as
; well as to incoming calls... be careful what you put here.
include = extens-acme
exten = i,1,Playback(pbx-invalid)
exten = i,n,Goto(s,exten)
exten = t,1,Goto(s,goodbye)
[internal-acme]
exten = s,1,Answer()
exten = s,n(exten),Background(vm-enter-num-to-call)
exten = s,n,WaitExten(5)
exten = s,n(goodbye),Playback(vm-goodbye)
exten = s,n(end),Hangup()
include = outbound-toll
include = outbound-local
include = extens-acme
; for our SIP phones, we can program a non-numeric extension
exten = voicemail,1,VoicemailMain([EMAIL PROTECTED])
exten = voicemail,n,Hangup()
; and for DTMF coming through an ATA...
exten = 777,1,Goto(voicemail)
[extens-acme]
exten = 111,1,Macro(stdexten,111,${PHILIP})
exten = 111,n,Goto(s,exten)
...
[outbound-local]
exten = _NXX,1,Dial(${TRUNK}/${AREA}${EXTEN},,r)
exten = _NXX,n,Congestion()
exten = _NXX,n,Hangup()
[outbound-toll]
exten = _NX,1,Dial(${TRUNK}/${EXTEN},,r)
exten = _NX,n,Congestion()
exten = _NX,n,Hangup()
exten = _011.,1,Dial(${TRUNK}/${EXTEN:3},,r)
exten = _011.,n,Congestion()
exten = _011.,n,Hangup()
Note: we had to modify the stdexten macro to be:
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well)
;
[asterisk-users] T.38 fax solution, opinions?
Hi, I'm putting together a fax solution for my company that utilizes T.38. I wanted to throw out my plan and get some feedback if anyone is doing something similar or sees a blatant problem with it. We're currently rolling out SPA-942 phones for the standard desk phone with vanilla Asterisk 1.4.15 (just upgraded it today) on the back end. Most calls for satellite offices are handled by VoIP providers (for voice Vitelity inbound, Gafachi outbound). These satellite offices are using a T.38 fax DID from Gafachi, passed through the Asterisk server to a Linksys 3102 ATA and then to a POTS fax machine. This all works well thus far. Our HQ has a full voice PRI, terminated on the Asterisk server with a TE120P. Additionally, right now they have five fax lines totally separate from the PRI that are used for POTS fax machines. I'm thinking of porting those numbers to the PRI and purchasing a TDM880B (comes with eight FXS modules) and routing the fax DIDs to the 880 in Asterisk. Five of the ports would connect into a Linksys 3102 that would speak T.38 to what would be our new fax environment (Exchange 2007 Unified Messaging). That part isn't implemented yet, but it shouldn't be a problem. Once it's implemented I'll probably reroute the Gafachi T.38 fax DIDs to Exchange through Asterisk (with sipX in there somewhere). The part(s) I'm unsure about is the TDM880B. I haven't used a FXS card with Asterisk, and I certainly haven't used a fax machine on that FXS. Additionally, I'm not 100% sure the 3102 will talk directly to Exchange UM yet, but that's something I can figure out myself soon; I'm just not sure about spending the cash for a TDM880B without knowing someone has thrown faxes through it from a PRI terminated on the same box from a separate card. Anyway, thoughts, criticisms, insults and stinging barbs all welcome. ___ --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
Re: [asterisk-users] Open Asterisk Exchange Project
From my point of view, today's IMAP implementation in Asterisk 1.4 is unusable (see http://bugs.digium.com/view.php?id=10487). I still have to meet anyone using it daily, without issues. 2007/12/9, Dan Austin [EMAIL PROTECTED]: Tzafrir wrote: On Sun, Dec 09, 2007 at 03:30:13PM -0500, Michelle Dupuis wrote: Well, we can already integrate to major platforms via SMTP. The real value is in deep integration to the most popular email platform in business: Exchange. I would love to see smart Exchange integration, where deleting the VM attached email will delete the corresponding message from asterisk. My clients would eat that up. IMAP support for voicemail? That would be an option if Microsoft had not implimeted such a limited set of IMAP features in Exchange. Exchange does not support the concept of master-user This master-user was dropped in Exchange 2007. when using IMAP, so you'd need to have the password of every Exchange account you wanted to integrate with Asterisk voicemail. I suppose that the addition of IMAP support to Asterisk's voicemail application might open the way for someone to look at using a MAPI library, such as the one being developed by the OpenChange folks. Dan ___ --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 ___ --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
