[asterisk-users] Segmenting A Configration File
Hi List, I am planning a multi-tenant VoIP services system with Asterisk, using configuration tweaks. Having all the tenant configurations in one configuration file is overwhelming. I would like to segment the configuration files and include them in the main configuration file. Is it possible? For e.g. I would like to have the main extenstions.conf file to include tenant01_extenstions.conf, tenant02_extensions.conf. By this way it is easy to manage the configurations of each tenant. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Segmenting A Configration File
Sure, you can include multiple files from the general extension.conf. You can do the same for the sip.conf. Leandro I am typing from my mobile phone... Il giorno 11/ago/2012 12:17, Kannan vasdevelo...@gmail.com ha scritto: Hi List, I am planning a multi-tenant VoIP services system with Asterisk, using configuration tweaks. Having all the tenant configurations in one configuration file is overwhelming. I would like to segment the configuration files and include them in the main configuration file. Is it possible? For e.g. I would like to have the main extenstions.conf file to include tenant01_extenstions.conf, tenant02_extensions.conf. By this way it is easy to manage the configurations of each tenant. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Segmenting A Configration File
On 12-08-11 06:16 AM, Kannan wrote: Hi List, I am planning a multi-tenant VoIP services system with Asterisk, using configuration tweaks. Having all the tenant configurations in one configuration file is overwhelming. I would like to segment the configuration files and include them in the main configuration file. Is it possible? For e.g. I would like to have the main extenstions.conf file to include tenant01_extenstions.conf, tenant02_extensions.conf. By this way it is easy to manage the configurations of each tenant. I've been playing a lot with the following configuration layout[1], so far I am pretty happy how well it is working. We define the default values we want for the default configuration files, then add the site specific settings under the specific directory. Like I said, works very well for us. [1] https://github.com/kickstandproject/asterisk/tree/master/debian/ast_config -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Segmenting A Configration File
On Sat, Aug 11, 2012 at 3:16 AM, Kannan vasdevelo...@gmail.com wrote:
I am planning a multi-tenant VoIP services system with Asterisk, using
configuration tweaks. Having all the tenant configurations in one
configuration file is overwhelming. I would like to segment the
configuration files and include them in the main configuration file. Is it
possible?
For e.g. I would like to have the main extenstions.conf file to include
tenant01_extenstions.conf, tenant02_extensions.conf. By this way it is easy
to manage the configurations of each tenant.
We put each tenant's sip and extensions config files in
/etc/asterisk/accounts and then do an include for that directory in the
main files.
We keep all the voicemail.conf in one because changes to passwords will NOT
be saved to included files. We used to use includes for voicemail but that
meant no password changes.
The main file has a list of all phone numbers in the system in numerical
order where we set the account name, and then we send them to the proper
context like this:
exten = 12015551212,1,Set(CDR(accountcode)=johnsmith)
exten = _X.,n(cont),Goto(${CDR(accountcode)}#did,${EXTEN},1)
There's a bunch of other stuff in there where we do line counting and such.
--
Carlos Alvarez
TelEvolve
602-889-3003
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Segmenting A Configration File
On 12-08-11 11:10 AM, Carlos Alvarez wrote: On Sat, Aug 11, 2012 at 3:16 AM, Kannan vasdevelo...@gmail.com wrote: I am planning a multi-tenant VoIP services system with Asterisk, using configuration tweaks. Having all the tenant configurations in one configuration file is overwhelming. I would like to segment the configuration files and include them in the main configuration file. Is it possible? For e.g. I would like to have the main extenstions.conf file to include tenant01_extenstions.conf, tenant02_extensions.conf. By this way it is easy to manage the configurations of each tenant. We put each tenant's sip and extensions config files in /etc/asterisk/accounts and then do an include for that directory in the main files. We keep all the voicemail.conf in one because changes to passwords will NOT be saved to included files. We used to use includes for voicemail but that meant no password changes. This is no longer the case. Starting with 1.8 a new voicemail.conf setting (passwordlocation) has been added[1] to allow you to store the passwords outside the voicemail.conf file. With this setting the password gets written to secret.conf file within spooldir for each mailbox. That way, you can then breakout each mailbox into separate config files with include statements. [1] http://svnview.digium.com/svn/asterisk?revision=225406view=revision -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple channel for SIP users
Hi Team, I use Asterix 1.6.2.9-2 what is running on debian squeeze. I completely statisfied this software. I did everything I want so far. I love it so much, but there is a point where I can not step through. 1) I have connected to my telephone provider as a SIP client, but my Asterisk only one call make to the world in same time. My provider does not limit the number of simultaneous calls. The only limit is the bandwidth of my local internet link. How can I configure my asterisk to create more than one simultaneous calls through my provider? 2) If I use an ATA, which has 2 SIP clients. These SIP clients is the same asterisk user, but asterisk register only the last one. May I got chance for registering ATA with the same users in the asterisk or every ATA must have two different asterisk user for working well? Thanks for any hints in advance! Best regards, Gabor Hatos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple channel for SIP users
2012/8/11 Hatos Gabor ha...@ggki.hu Hi Team, I use Asterix 1.6.2.9-2 what is running on debian squeeze. I completely statisfied this software. I did everything I want so far. I love it so much, but there is a point where I can not step through. 1) I have connected to my telephone provider as a SIP client, but my Asterisk only one call make to the world in same time. My provider does not limit the number of simultaneous calls. The only limit is the bandwidth of my local internet link. How can I configure my asterisk to create more than one simultaneous calls through my provider? Asterisk has no limitation on the number of simultaneous calls. Just place another call while one call is already going... 2) If I use an ATA, which has 2 SIP clients. These SIP clients is the same asterisk user, but asterisk register only the last one. May I got chance for registering ATA with the same users in the asterisk or every ATA must have two different asterisk user for working well? Ata I have found so far allows to set two distinct SIP account for each one of the FXS/FXO ports they have. Leandro Thanks for any hints in advance! Best regards, Gabor Hatos -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
And where there is a 'government' involved, bypassing their restrictions may have serious consequences. Yup, true. Since, I'm a single user *experimenting* with PBXs, I'm hopefully out of danger zone. But yes, when there is a government is almost no way. :D -- Sincerely, Sazzad Bin Kamal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
I have been observing that Twinkle send same value in both *from* and *to*fields. Like the following: from/to: account_name@asterisk_host:port_no Shouldn't it be: from : account_name@twinkle_host:port_no to : account_name@asterisk_host:port_no ?? Also, I realized there is not need to create a single SIP packet with REGISTER message. Twinkle is already dong that. What probably, I should do is copy-paste the log, save them is a file, feed it to other programs and tinker with them. Any hint for a noob? -- Sincerely, Sazzad Bin Kamal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
On Sun, 12 Aug 2012, Sazzad wrote: Also, I realized there is not need to create a single SIP packet with REGISTER message. Twinkle is already dong that. What probably, I should do is copy-paste the log, save them is a file, feed it to other programs and tinker with them. Any hint for a noob? Both sipp and sipsak can send REGISTER packets. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
Both sipp and sipsak can send REGISTER packets. Thanks. sipp was successfully built on my Debian Squeeze. But when I run it, I get errors. The error was originating from a function named 'xp_get_value'. So I thought it might be related to Windows XP and its windows specific. Am I wrong? Any experience with it on Linux? Haven't tried sipsak yet. Luckily, now I'm getting debugging infohttp://asterisk.pastebin.ca/2179329at Asterisk console. The pattern is, every-time I get this info, I have to, at least, restart Twinkle and/or Asterisk. If I don't, no matter how many times I issue 'register' command, it doesn't work. -- Sincerely, Sazzad Bin Kamal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Segmenting A Configration File
Hi
Have you seen thirdlane?
Thirdlane has a multitenant version.
Regards
On Aug 11, 2012 11:11 AM, Carlos Alvarez car...@televolve.com wrote:
On Sat, Aug 11, 2012 at 3:16 AM, Kannan vasdevelo...@gmail.com wrote:
I am planning a multi-tenant VoIP services system with Asterisk, using
configuration tweaks. Having all the tenant configurations in one
configuration file is overwhelming. I would like to segment the
configuration files and include them in the main configuration file. Is it
possible?
For e.g. I would like to have the main extenstions.conf file to include
tenant01_extenstions.conf, tenant02_extensions.conf. By this way it is easy
to manage the configurations of each tenant.
We put each tenant's sip and extensions config files in
/etc/asterisk/accounts and then do an include for that directory in the
main files.
We keep all the voicemail.conf in one because changes to passwords will
NOT be saved to included files. We used to use includes for voicemail but
that meant no password changes.
The main file has a list of all phone numbers in the system in numerical
order where we set the account name, and then we send them to the proper
context like this:
exten = 12015551212,1,Set(CDR(accountcode)=johnsmith)
exten = _X.,n(cont),Goto(${CDR(accountcode)}#did,${EXTEN},1)
There's a bunch of other stuff in there where we do line counting and such.
--
Carlos Alvarez
TelEvolve
602-889-3003
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] best free fax solution with asterisk
On 8/11/2012 8:05 AM, virendra bhati wrote: Hi team, I want to configure fax with asterisk. there a lot of fax link i found by google but not working perfectly. my setup as follow asterisk 10.x centos 5.8 Want to used T.38 with SpanDSP... Please suggest me the best way. and how to test FoIP ? I use Asterisk 10.3.1, SpanDSP 0.0.6, and Ubuntu 11.10 connecting to Gafachi.com. It works with probably 95% success rate talking via T.38. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
On Sunday 12 Aug 2012, Steve Edwards wrote: On Sat, 11 Aug 2012, SamyGo wrote: It takes a VPN or in near future WebRTC(in other words Knowledge) to become one powerful guy. With these technologies you don't need to care what your ISP or govt. is blocking. Where there is will, there are ways. And where there is a 'government' involved, bypassing their restrictions may have serious consequences. It's not necessarily a Government thing. In India, some ISPs -- who are also telcos -- have unilaterally blocked 5060/UDP traffic to prevent VoIP eating into their PSTN business. Of course, India has some retrograde VoIP rules, but blocking 5060/UDP isn't an official requirement. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Segmenting A Configration File
On Saturday 11 Aug 2012, Kannan wrote: I am planning a multi-tenant VoIP services system with Asterisk, using configuration tweaks. Having all the tenant configurations in one configuration file is overwhelming. I would like to segment the configuration files and include them in the main configuration file. Is it possible? For e.g. I would like to have the main extenstions.conf file to include tenant01_extenstions.conf, tenant02_extensions.conf. By this way it is easy to manage the configurations of each tenant. We have developed a completely parametrised solution for one client, where she can configure contexts without ever having to touch the main Asterisk files. For each context, the dialplan checks configuration values for recording, permitting calls to various types of extensions, adding to queues, barge-in, etc and enables or disables those services depending on the parameters provided. You can even create custom extensions and invoke AGIs at runtime if you need more fine-tuning. All these customisations -- the client's configuration, the dialplan functions, users, etc. are in separate files, #included by the main Asterisk configurations. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 1.0.4 Released
The AstLinux Team is happy to announce the release of 1.0.4. New in this release: -- Asterisk 1.4.44 and 1.8.14.1 -- DAHDI, dahdi-linux 2.6.1 and dahdi-tools 2.6.1 -- wanpipe, version bump to 3.5.27 -- rhino, version bump to 0.99.6b2. Support is now enabled again by default. -- libPRI, upstream patch to add layer 2 persistence option to customize the layer 2 behavior on BRI PTMP lines. (Thanks to Michael Keuter) -- PHP version bump to 5.3.14 to address security issues. -- Security fixes for OpenSSL -- miniupnpd added (disabled by default) to support Universal Plug and Play. (Many thanks to David Kerr) -- mtr added. Network diagnostic tool that combines ping and traceroute. -- Updates to the web interface including the addition of a MeetMe tab, firewall enhancements and UPnP support. For the complete changelog and to download the install images go to the following pages: http://www.astlinux.org/release/104-asterisk-18141 http://www.astlinux.org/release/104-asterisk-1444 The AstLinux Team -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
