Re: [asterisk-users] Managing complex setups with Asterisk
On Friday 16 Nov 2012, martin f krafft wrote: also sprach Paul Belanger paul.belan...@polybeacon.com [2012.11.08.2304 +0100]: Either way, it sounds like you need to store your data some place and start building it out. To recap: given that Asterisk RealTime doesn't really provide anything more than real-time access to data (i.e. the data in the database are not any more structured that they are in /etc/asterisk), any more logical and/or abstract approach to Asterisk configuration means that the data have to come from elsewhere and be brought into shape. Either the abstraction happens in a relational database and Asterisk accesses stored procedures or views (I would not use LDAP due to childhood traumata), or the relational database is used to generate Asterisk's configuration files, or some other data source is used to generate these configuration files. It's a shame that noone has done anything into this direction yet. On the other hand, it means that there aren't already a dozen PHP+MySQL hacks out there, and that's a good thing. So if I design the database (PostgreSQL), anyone interested in providing a frontend, e.g. using Django? Are people interested in discussing the design here and making it widely usable? I only have my own three use-cases to refer to, and I would probably impose my own paradigms… Does anyone already have something done into that domain? Interesting. Let's discuss. Warning: Not a fan of using whitespace as semantic markup, so no Django this side. Fine with Perl or Java, though. 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] Conf into a call in progress
On 15 Nov 2012, at 15:44, Michael wrote: Hi Aldo, Thank you very much for answering my question. Can you kindly elaborate on how to do the following or at least where to read about the way to do it? Hi Michael, sure... I am sending you -by direct mail- a diagram that tries to illustrate what I would try to do. (I do not know if this list allows attachments; generally it's not permitted...). send both channels of the active call 111 - 22334455 to a context that joins them in a conference room. AMI has a useful command for that task: Redirect, see here: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Redirect and here: http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+Transfer If you are manipulating a call not from one of the connected terminals (e.g. your phone) you have to take care of both channels. This is what the Redirect command does. It lets you specify what to do with both channels: they can be sent to the same context or each to one context by itself. Finally you are able to make changes on one channel only... through AMI, I would originate the call to 22556677 and join it into the conference. So the plan would be to first send the two channels to a conference room (an ad hoc one), using a first redirect command. This is made to get the conf. room where the three way call will take place AND to be able to call the third party without losing the original call partner's channel... A second Redirect command should detach the user's channel from the conference and send it to a context that connects him/her to the third party, letting the original user offer the 3 way call. If the call is accepted, than a third redirect would send both channels to the conference room created at step 1, where the other party is waiting... The dynamic conference is closed either by the original call party hanging up his/her channel or with a direct AMI hangup command doing the same thing. Clearly this is logically equivalent to a manual transfer of the user's call party into a conference room. Then calling the second call party and transfer him/her to the conference and seeing the user finally dialing him/herself into the conference. You can do that with AMI, provided you have some means to make some sort of UI for the whole process... Thank you very much, Michael You're welcome: hth! Aldo -- _ -- 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] Managing complex setups with Asterisk
also sprach Raj Mathur (राज माथुर) r...@linux-delhi.org [2012.11.16.1005 +0100]: Warning: Not a fan of using whitespace as semantic markup, so no Django this side. Fine with Perl or Java, though. As long as we can agree on using a database (i.e. no MySQL) or the filesystem (Git…), then the question of which language to use for a frontend is secondary. I wouldn't chose Java myself, but I suspect that the job is enough text processing that Perl would actually be a sensible choice — except I won't help since I don't know it well. But shouldn't the first step be a mixture of database design and requirement specification? I would like a solution that keeps users, sites, and numbers (belonging to trunks (hardware, as well as SIP)) separate and then basically allows for free combinations. User A might have a desk at site I, to which a range of numbers is assigned, and in addition to an internal number (e.g. a one digit site prefix followed by a two digit number, or a site-independent number assigned per person), one of those externals rings at A's desk. User B might roam between sites I and II and either should have the same internal/external numbers ringing at both desks, or require some sort of login to let the system know where to ring. User C might have a desk with a phone at site II, but is out most of the time, and calls should also ring on his/her cell. User D has a smart phone and wants both his desk and the smart phone to ring. All users want voicemail and be able to configure the time until voicemail answers. During vacation etc., a forwarding number should be configurable. Some users might want their voicemail to say e.g. press 1 now to be transferred to my cell. We would also want to be able to specify per-user whether to use UDP, TCP or IAX, who can transfer and park calls, who can record them with mix monitor, who can create ad-hoc conferences, their language, who has a video telephone… … and of course there ought to be a way to set user-specific sip.conf settings. On top, it would be nice if there were some sort of group inheritance. This sounds a bit like LDAP, except LDAP can't actually do it. What I mean is that I'd really like to define a group of e.g. managers who all have internal numbers beginning with 11 and secretaries who can create conferences, and then associate users with (multiple) groups, inheriting and merging the settings. These are — I think — my base requirements. What would you add? -- martin | http://madduck.net/ | http://two.sentenc.es/ quick!! act as if nothing has happened! spamtraps: madduck.bo...@madduck.net digital_signature_gpg.asc Description: Digital signature (see http://martin-krafft.net/gpg/sig-policy/999bbcc4/current) -- _ -- 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] Intruder
I am in the asterisk CLI and can see an unidentified caller trying the make calls out of the asterisk system. How do I stop them? How do I identify them and how can I see how the go in? This is an example of what I would see: NOTICE[4098]: chan_sip.c:20063 handle_request_invite: Call from '' to extension '90111235551212' rejected because extension not found. Felix This electronic message contains information from BOSH Global Services which may be company sensitive, proprietary, privileged or otherwise protected from disclosure. The information is intended to be used solely by the recipient(s) named above. If you are not an intended recipient, be aware that any review, disclosure, copying, distribution or use of this transmission or its contents is prohibited. If you have received this transmission in error, please notify the sender immediately. -- _ -- 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] Reminder: Asterisk 10 Support Window Ending
Hello! This is a friendly reminder that the support window for bug fixes for Asterisk 10 will come to an end in one month. After 12-15-2012, Asterisk 10 will receive security fixes for an additional year, with its full EOL occurring on 12-15-2013. Users of Asterisk 10 are encouraged to move to the next major version, Asterisk 11, at their earliest convenience. Asterisk 11 is an LTS release and is supported through 10-25-2016, with its full EOL occurring on 10-25-2017. For more information on Asterisk versions and their supported lifetimes, please see the following wiki page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions Thanks you for your continued support of Asterisk! -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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] Intruder
Hi Felix, you have several things to check: netstat -a -n --udp --tcp will show you connections and connection attempts on network layer level. You have to look for incoming connections to port 5060 and if the call has been established for connections on your rtp ports. (see rtp.conf). If you can see connections not supposed to be there: thats your intruder ;-) I suggest you disable guest calls and you configure a default context in which dialed extensions can't be routed to charged destinations. sip.conf: allowguests=no defaultcontext=default extensions.conf: [default] exten = _X.,1,Answer() exten = _X.,n,PlayBack(silence/1) exten = _X.,n,PlayBack(ss-noservice) exten = _X.,n,PlayBack(silence/1) exten = _X.,n,MusicOnHold(default,10) exten = _X.,n,PlayBack(silence/1) exten = _X.,n,PlayBack(vm-goodbye) exten = _X.,n,HangUp() The next step would be using fail2ban or something similiar to check the asterisk log for intruders. fail2ban recognized them and dynamically sets appropriate firewall rules. Good luck. best regards, Ruben Am 16.11.2012 17:20, schrieb Felix Vazquez: I am in the asterisk CLI and can see an unidentified caller trying the make calls out of the asterisk system. How do I stop them? How do I identify them and how can I see how the go in? This is an example of what I would see: NOTICE[4098]: chan_sip.c:20063 handle_request_invite: Call *from '' *to extension '90111235551212' rejected because extension not found. Felix This electronic message contains information from BOSH Global Services which may be company sensitive, proprietary, privileged or otherwise protected from disclosure. The information is intended to be used solely by the recipient(s) named above. If you are not an intended recipient, be aware that any review, disclosure, copying, distribution or use of this transmission or its contents is prohibited. If you have received this transmission in error, please notify the sender immediately. -- _ -- 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] Intruder
Hi Felix, ngrep -W byline port 5060|grep -B1 INVITE sip Markus Am 16.11.2012 17:50, schrieb Ruben Rögels: Hi Felix, you have several things to check: netstat -a -n --udp --tcp will show you connections and connection attempts on network layer level. You have to look for incoming connections to port 5060 and if the call has been established for connections on your rtp ports. (see rtp.conf). If you can see connections not supposed to be there: thats your intruder ;-) I suggest you disable guest calls and you configure a default context in which dialed extensions can't be routed to charged destinations. sip.conf: allowguests=no defaultcontext=default extensions.conf: [default] exten = _X.,1,Answer() exten = _X.,n,PlayBack(silence/1) exten = _X.,n,PlayBack(ss-noservice) exten = _X.,n,PlayBack(silence/1) exten = _X.,n,MusicOnHold(default,10) exten = _X.,n,PlayBack(silence/1) exten = _X.,n,PlayBack(vm-goodbye) exten = _X.,n,HangUp() The next step would be using fail2ban or something similiar to check the asterisk log for intruders. fail2ban recognized them and dynamically sets appropriate firewall rules. Good luck. best regards, Ruben Am 16.11.2012 17:20, schrieb Felix Vazquez: I am in the asterisk CLI and can see an unidentified caller trying the make calls out of the asterisk system. How do I stop them? How do I identify them and how can I see how the go in? This is an example of what I would see: NOTICE[4098]: chan_sip.c:20063 handle_request_invite: Call *from '' *to extension '90111235551212' rejected because extension not found. Felix This electronic message contains information from BOSH Global Services which may be company sensitive, proprietary, privileged or otherwise protected from disclosure. The information is intended to be used solely by the recipient(s) named above. If you are not an intended recipient, be aware that any review, disclosure, copying, distribution or use of this transmission or its contents is prohibited. If you have received this transmission in error, please notify the sender immediately. -- _ -- Bandwidth and Colocation Provided byhttp://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 -- _ -- 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] Intruder
- Original Message - From: Felix Vazquez felix.vazq...@theboshgroup.com To: asterisk-users@lists.digium.com Sent: Friday, November 16, 2012 11:20:46 AM Subject: [asterisk-users] Intruder I am in the asterisk CLI and can see an unidentified caller trying the make calls out of the asterisk system. How do I stop them? How do I identify them and how can I see how the go in? This is an example of what I would see: NOTICE[4098]: chan_sip.c:20063 handle_request_invite: Call from '' to extension '90111235551212' rejected because extension not found. I would recommend you read README-SERIOUSLY.bestpractices.txt, top level of source code. Another thing you can do is turn on security logging if you are using Asterisk 10/11. Take a look at logger.conf. It may provide you with some extra information on who is trying to make the call. Take a look at this page: https://wiki.asterisk.org/wiki/display/AST/Important+Security+Considerations I would recommend using fail2ban as well. Michael (elguero) -- _ -- 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] Intruder
I created my own Whitelist and Blacklist system. When I make an
outgoing call, the number is automatically added to my Whitelist
database and I can add numbers to the Blacklist manually or by pressing
the *.
You can use this for incoming/outgoing calls however you want to setup
your extensions.
If a Whitelisted caller is calling, I change the Caller(name) =
Whitelist so I know it's ok to answer.
If a Blacklisted caller is calling, I play a message and hangup.
I get a lot of 8** calls from solicitors so here is my dialplan and
database:
I pass the call to these Macros before it reaches anyone and I can block
calls by date time too.
Mysql Blacklist Database
blacklistid, callerid_from, callerid_to, description, times, days,
months, playback
35, '%8775160592', '%', 'Solicitor keeps calling, '*', '*', '*',
'discon-or-out-of-service'
32, '%', '%2134271', 'Kids Friends cant call after midnight and before
8am', '00:00-08:00', '*', '*',
'sorry-cant-let-you-do-that2please-try-again-later'
[trunk]
..
exten = _X!,n,Macro(blacklist,${CALLERID(num)},${EXTEN})
exten = _X!,n,Macro(whitelist,${CALLERID(num)},${EXTEN})
exten = _X!,n,Set(DB(global/lastcallerid)=${CALLERID(num)})
exten = _X!,n,Goto(incoming,start,1)
[macro-blacklist]
exten = s,1,MYSQL(Connect connid ${db_host} ${db_user} ${db_pass}
${db_name})
exten = s,n,MYSQL(Query resultid ${connid} SELECT blacklistid,
callerid_from, callerid_to, times, days, months, playback FROM blacklist
WHERE '${ARG1}' LIKE callerid_from AND '${ARG2}' LIKE callerid_to)
exten = s,n,MYSQL(Fetch fetchid ${resultid} blacklistid callerid1
callerid2 times days months playback)
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})
exten = s,n,GoToIf($[${blacklistid} = ]?call,1:time,1)
exten = time,1,GotoIfTime(${times},${days},${months}?fail,1:call,1)
exten = fail,1,NoOp(Blacklisted ${callerid1} to ${callerid2})
exten = fail,n,GoTo(blacklisted,s,1)
exten = call,1,NoOp(Not Blacklisted ${ARG1} to ${ARG2})
[macro-blacklist-add]
exten = s,1,MYSQL(Connect connid ${db_host} ${db_user} ${db_pass}
${db_name})
exten = s,n,MYSQL(Query resultid ${connid} INSERT IGNORE INTO blacklist
(callerid_to, callerid_from, description) VALUES
('${ARG1}','${ARG2}','Blacklisted'))
exten = s,n,MYSQL(Disconnect ${connid})
[macro-whitelist]
exten = s,1,MYSQL(Connect connid ${db_host} ${db_user} ${db_pass}
${db_name})
exten = s,n,MYSQL(Query resultid ${connid} SELECT whitelistid,
callerid_from, callerid_to, description FROM whitelist WHERE '${ARG1}'
LIKE callerid_from AND '${ARG2}' LIKE callerid_to)
exten = s,n,MYSQL(Fetch fetchid ${resultid} whitelistid callerid1
callerid2 description)
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})
exten = s,n,GoToIf($[${whitelistid} = ]?not,1:is,1)
exten = is,1,NoOp(Whitelisted ${ARG1} to ${ARG2})
exten = is,n,Set(CALLERID(name)=${description})
exten = not,1,NoOp(Not Whitelisted ${ARG1} to ${ARG2})
exten = not,n,Set(CALLERID(name)=Unknown)
[macro-whitelist-add]
exten = s,1,MYSQL(Connect connid ${db_host} ${db_user} ${db_pass}
${db_name})
exten = s,n,MYSQL(Query resultid ${connid} INSERT IGNORE INTO whitelist
(callerid_to, callerid_from) VALUES ('%','${ARG2}'))
exten = s,n,MYSQL(Disconnect ${connid})
[blacklisted]
exten = s,1,Set(CALLERID(name)=Blacklisted)
exten = s,n,Wait(3)
exten = s,n,Playback(${playback})
exten = s,n,HangUp()
If you want to add a KEY to your dialplan to add to blacklist or whitelist:
[roy]
exten = roy,*,Macro(blacklist-add,%,${DB(global/lastcallerid)})
exten = roy,#,Macro(whitelist-add,%,${DB(global/lastcallerid)})
Co-op Vacation Rentals
www.coopvr.com
15218 Summit Ave
Suite #300-354
Fontana, CA 92336
Phone/Fax (855) 760-COOP (2667)
On 11/16/2012 8:20 AM, Felix Vazquez wrote:
I am in the asterisk CLI and can see an unidentified caller trying the
make calls out of the asterisk system. How do I stop them? How do I
identify them and how can I see how the go in?
This is an example of what I would see:
NOTICE[4098]: chan_sip.c:20063 handle_request_invite:
Call *from '' *to extension '90111235551212' rejected because
extension not found.
Felix
This electronic message contains information from BOSH Global Services
which may be company sensitive, proprietary, privileged or otherwise
protected from disclosure. The information is intended to be used
solely by the recipient(s) named above. If you are not an intended
recipient, be aware that any review, disclosure, copying, distribution
or use of this transmission or its contents is prohibited. If you have
received this transmission in error, please notify the sender immediately.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
[asterisk-users] Pager Duty Service on Asterisk
Hi, Does Asterisk has pager duty feature and write ups or How To's to setup? Regards, Kaushal -- _ -- 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] Pager Duty Service on Asterisk
You can accomplish this with time conditions. On Nov 16, 2012 7:50 PM, Kaushal Shriyan kaushalshri...@gmail.com wrote: Hi, Does Asterisk has pager duty feature and write ups or How To's to setup? Regards, Kaushal -- _ -- 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] Pager Duty Service on Asterisk
On Sat, Nov 17, 2012 at 7:22 AM, Jared Baxley jared.bax...@gmail.comwrote: You can accomplish this with time conditions. Thanks Jared. Any docs or tutorials to refer to set up? Regards, Kaushal -- _ -- 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] Pager Duty Service on Asterisk
A google search will yield dozens of how to guides fot asterisk time conditions. ... but your version and specific deployment must me taken into account. If you are looking for someone to implement this for you feel free to contact me. Jared Baxley 205.292.0744 On Nov 16, 2012 7:54 PM, Kaushal Shriyan kaushalshri...@gmail.com wrote: On Sat, Nov 17, 2012 at 7:22 AM, Jared Baxley jared.bax...@gmail.comwrote: You can accomplish this with time conditions. Thanks Jared. Any docs or tutorials to refer to set up? Regards, Kaushal -- _ -- 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] Managing complex setups with Asterisk
On Friday 16 Nov 2012, martin f krafft wrote: also sprach Raj Mathur (राज माथुर) r...@linux-delhi.org [2012.11.16.1005 +0100]: Warning: Not a fan of using whitespace as semantic markup, so no Django this side. Fine with Perl or Java, though. As long as we can agree on using a database (i.e. no MySQL) or the filesystem (Git…), then the question of which language to use for a frontend is secondary. I wouldn't chose Java myself, but I suspect that the job is enough text processing that Perl would actually be a sensible choice — except I won't help since I don't know it well. But shouldn't the first step be a mixture of database design and requirement specification? I would like a solution that keeps users, sites, and numbers (belonging to trunks (hardware, as well as SIP)) separate and then basically allows for free combinations. User A might have a desk at site I, to which a range of numbers is assigned, and in addition to an internal number (e.g. a one digit site prefix followed by a two digit number, or a site-independent number assigned per person), one of those externals rings at A's desk. User B might roam between sites I and II and either should have the same internal/external numbers ringing at both desks, or require some sort of login to let the system know where to ring. User C might have a desk with a phone at site II, but is out most of the time, and calls should also ring on his/her cell. User D has a smart phone and wants both his desk and the smart phone to ring. All users want voicemail and be able to configure the time until voicemail answers. During vacation etc., a forwarding number should be configurable. Some users might want their voicemail to say e.g. press 1 now to be transferred to my cell. We would also want to be able to specify per-user whether to use UDP, TCP or IAX, who can transfer and park calls, who can record them with mix monitor, who can create ad-hoc conferences, their language, who has a video telephone… … and of course there ought to be a way to set user-specific sip.conf settings. On top, it would be nice if there were some sort of group inheritance. This sounds a bit like LDAP, except LDAP can't actually do it. What I mean is that I'd really like to define a group of e.g. managers who all have internal numbers beginning with 11 and secretaries who can create conferences, and then associate users with (multiple) groups, inheriting and merging the settings. These are — I think — my base requirements. What would you add? I'll talk to clients and get a feature list from them too. Then we can filter into initial, advanced and nice to have categories. Unless enough other people are interested (yes, asking on Saturday morning is a good way of ensuring no one answers :) , we ought to take this to private mail. 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
