Re: [asterisk-users] Queue_log into MySQL - best practices
On Sunday 25 Nov 2012, Dmitry wrote: [snip] 3) Still know nothing about odbc support for queue_log Happily using ODBC (PostgreSQL, but should be mostly DB-independent) for queue_log here. The setup is a bit hairy, but can share if enough people show interest. 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] 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] 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
[asterisk-users] Fully utilise all PRIs in a DAHDI group
Hi, Our client has DAHDI groups with 4 PRIs in each group (one 4-port interface per group), up to 6 groups per server. When we dial, we can specify the group to be used for dialling, and our dial plan automatically distributes calls over multiple servers and multiple groups within a server. The way Asterisk dials by default is to use the lowest-numbered free line in a group to place a call. This is technically fine. However, what it means for our client is that the first couple of PRIs in a group tend to get the bulk of calls, the other two remain more-or-less unutilised. This is a problem, since there are call commitments to the Telco for each PRI line. The Telco tends to get all soggy and hard to light if some of the PRIs are used way below committed call levels. One solution is to group at the individual PRI level, so the load balancing automatically takes care of fair utilisation of each PRI. However, for various reasons we'd prefer not to do this. Another solution would be if Asterisk could choose a random (or LRU or LCU or round-robin or any other scheme) PRI within a group when dialling. Any roughly fair way to distribute calls to PRIs within a DAHDI group would be fine. Is there some way to achieve this? Asterisk 1.8.8 on Debian Squeeze. 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] Fully utilise all PRIs in a DAHDI group
On Wednesday 17 Oct 2012, Tony Mountifield wrote: In article 201210171813.45334.r...@linux-delhi.org, Raj Mathur (à€°à€Ÿà€ à€®à€Ÿà€¥à¥ à€°) r...@linux-delhi.org wrote: Our client has DAHDI groups with 4 PRIs in each group (one 4-port interface per group), up to 6 groups per server. When we dial, we can specify the group to be used for dialling, and our dial plan automatically distributes calls over multiple servers and multiple groups within a server. The way Asterisk dials by default is to use the lowest-numbered free line in a group to place a call. This is technically fine. However, what it means for our client is that the first couple of PRIs in a group tend to get the bulk of calls, the other two remain more-or-less unutilised. This is a problem, since there are call commitments to the Telco for each PRI line. The Telco tends to get all soggy and hard to light if some of the PRIs are used way below committed call levels. One solution is to group at the individual PRI level, so the load balancing automatically takes care of fair utilisation of each PRI. However, for various reasons we'd prefer not to do this. Another solution would be if Asterisk could choose a random (or LRU or LCU or round-robin or any other scheme) PRI within a group when dialling. Any roughly fair way to distribute calls to PRIs within a DAHDI group would be fine. Is there some way to achieve this? Asterisk 1.8.8 on Debian Squeeze. Instead of dialling using DAHDI/g1/123456789, you can try using DAHDI/r1/123456789 to make Asterisk use the channels in round-robin order instead of always choosing the lowest free channel. See http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels (I could not find comparable information on the Asterisk WIKI at https://wiki.asterisk.org). Thanks to you and Steve Totaro, that's exactly what I need. 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] asterisk installation under a single directory
On Monday 15 Oct 2012, sudeep melekar wrote: hello, i want to install asterisk 1.8 in a single directory myasterisksetup i.e after asterisk installs it put some of it's installation files in different directories e.g /var/log/asterisk /var/run/asterisk and many more i want all this installation files to be under my directory myasterisksetup can any one provide me step by step installation of asterisk in a single directory i m completely new to asterisk so any help would be appreciated sh ./configure --prefix /home/sudeep/myasterisksetup 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] I can hear my own voice through the headset
On Thursday 04 Oct 2012, frangky robert wrote: Here is my IP-PBX setupmy setup is : sips softphone - asterisk - xorcom PSTN gateway - pstn line to telcoi'm using xlite for windows when I make a phone call (sip - outgoing channel),I can hear my own voice so clear. it's very annoying mewhen talking a little loud... any solution? Thanks, We've often faced this problem with SIP soft phones when the computer's sound system gain was set too high. You usually have to play around with microphone gain settings to get to the point where the echo disappears with the other party still being able to hear you. 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] Case-sensitivity of Dialplan variables.
On Tuesday 02 Oct 2012, Mark Michelson wrote: [snip] Some of you might be eager to propose a configuration option to decide which it should be. I'm sick of having hundreds of options in Asterisk to slightly tweak the behavior one way or another. This needs to go one way or the other, not be configurable. All dialplans that I've written so far will work fine in a case- sensitive environment. However, I appreciate that there will be legacy dialplans around that are, for one reason or another, case-inconsistent. To expect them all to switch to the new way of doing things immediately is impractical and unfair. So here's the proposal: make case-insensitivity a configuration option for one or two releases. Document the option (both externally and in the configuration file) with large warnings about how switching it on is DEPRECATED and how it will VANISH IN A FUTURE RELEASE. That will suit the people who do not wish to conform (they will not upgrade), the people who want to conform but need time (will have a few months to fix and test while still being able to use the latest Asterisk features) and the people who are already conformant (don't need to do anything). In short, my vote goes for case-sensitivity with a grace period for switching over. 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] Issue with PRI connection
On Sunday 23 Sep 2012, Ashish Agarwal wrote: vi*CLI dahdi show status Description Alarms IRQbpviol CRC Fra Codi Options LBO T4XXP (PCI) Card 0 Span 1OK 0 0 0 CCS HDB3 0 db (CSU)/0-133 feet (DSX-1) T4XXP (PCI) Card 0 Span 2OK 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) T4XXP (PCI) Card 0 Span 3OK 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) T4XXP (PCI) Card 0 Span 4OK 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) vi*CLI pri show spans PRI span 1/0: Down, Active PRI span 2/0: Down, Active PRI span 3/0: Down, Active PRI span 4/0: Down, Active A client had similar issues with providers in India until a wizard in their office figured out the correct cabling. Don't go by what the provider tells you, figure out the cable connections yourself. 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] Issue with PRI connection
On Monday 24 Sep 2012, Ashish Agarwal wrote: I have used 1 2 4 5 combination. Is that right? I wouldn't know, since I'm not the wizard :) But basically we had to do each provider's connections from scratch -- Airtel, BSNL, MTNL, Reliance, Tata. And as far as I recall, each provider had a different cable signalling scheme. Please don't top-post, it may get your posts ignored on the list. Use bottom- or inline-posting, and trim your replies. 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] Issue with PRI connection
On Monday 24 Sep 2012, Mitul Limbani wrote: Signalling frm remote side is down. Also just add crc4 in span 1,0,0 in dahdi/system.conf just like other spans. what is signalling= defined in your asterisk/chan_dahdi.conf ? Not necessarily. I guess you remember the problems we had in getting lines to work in Mumbai until we found the correct wire connections for each of our providers (Reliance and BSNL AFAIR). Eventually Raj Kumar (the wizard) did manage to figure it out and after that the lines worked just fine. 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] $agi-hangup() Does not hang up the channel
On Monday 17 Sep 2012, Mehdi Rahimi wrote: I need to use agi to handle some issue , after finishing agi i want to hang up the channel , if i call from an extension there is no problem but i want to be the same for PSTN (outside) caller , if someone call asterisk show the hang up channel but the caller is not disconnected and if meanwhile someone inside try to call from an extension the outide caller can listen to DTMF and everything . . . . I would be really grateful if you share your close experience . If this is on Analog (FXO) lines then you may be out of luck. In India, at least, the called party cannot hang up an incoming call on an analog line -- the caller has to do it. In any case, what happens if you execute HangUp() in your dialplan instead of in AGI? 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] $agi-hangup() Does not hang up the channel
On Monday 17 Sep 2012, Mehdi Rahimi wrote: Thank you for your reply i did it in both ways (AGI and DIALPLAN) but not working. so you mean it is because of telco ? what about digital lines such as E1 ? From my experience: the call gets disconnected if the called party executes HangUp on a digital line. The problem is only with Analog lines. 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] Digium AEX410, MTNL Mumbai Caller-ID problems
On Friday 14 Sep 2012, Patrick Lists wrote: On 09/14/2012 05:26 AM, Raj Mathur (राज माथुर) wrote: [snip] Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express) Your DAHDI and Asterisk versions are old so for starters I would update everything to the latest releases. See asterisk.org. I could compile the latest from source for the sake of testing, but will eventually have to move back to a packaged Asterisk for production. Since the Digium Debian Asterisk packages don't have a maintainer any more, that means production will eventually return back to 1.8.13. So if there's a good chance that the latest Asterisk and Dahdi packages will give better results in testing or might actually solve the problem, I'll be glad to compile from source. If not, then perhaps it's not worth polluting a production box with locally-compiled packages. 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] Digium AEX410, MTNL Mumbai Caller-ID problems
On Friday 14 Sep 2012, Richard Mudgett wrote: Continuing with the saga of Digium vs MTNL Mumbai, looking for suggestions on handling incoming Caller-ID issues. The card manages to grab a couple of (random) digits of the incoming CID, but they're more or less useless. Is there any way to fix this? Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express) chan_dahdi.conf contains: usecallerid = yes cidsignalling=dtmf cidstart=polarity_in Signalling is fxsks. Log (calling number is 9811066XXX): You appear to be suffering form https://issues.asterisk.org/jira/browse/ASTERISK-19610 It is fixed in the just released v1.8.17.0-rc1. Thanks, will test that on the weekend then. 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] MySQL Query : Calls Answered for 5 sec
On Friday 14 Sep 2012, RSCL Mumbai wrote: I am trying to construct MySQL query(s) to get a list of calls which lasted for less than 5 seconds between a given date range. Any help is appreciated. On the CDR database, to get all calls that lasted 5 seconds between 2012-09-01 and 2012-09-07 (inclusive), the MySQL query would be: select * from cdr where calldate = '2012-09-01' and calldate '2012-09-08' and duration 5; 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] Digium AEX410, MTNL Mumbai Caller-ID problems
On Friday 14 Sep 2012, Richard Mudgett wrote: Continuing with the saga of Digium vs MTNL Mumbai, looking for suggestions on handling incoming Caller-ID issues. The card manages to grab a couple of (random) digits of the incoming CID, but they're more or less useless. Is there any way to fix this? Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express) chan_dahdi.conf contains: usecallerid = yes cidsignalling=dtmf cidstart=polarity_in Signalling is fxsks. Log (calling number is 9811066XXX): You appear to be suffering form https://issues.asterisk.org/jira/browse/ASTERISK-19610 It is fixed in the just released v1.8.17.0-rc1. Outstanding! Just tried with 1.8.17.0-rc1 and the old Dahdi (2.5.0.1) and it delivered at least two CIDs just fine. Many thanks. Now to sort out the Asterisk packaged for Debian issue. Worst case, will package it myself :( 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] Digium AEX410, MTNL Mumbai Caller-ID problems
do_state_change: Changing state for DAHDI/1 - state 2 (In use) [Sep 14 08:21:14] DEBUG[9315]: devicestate.c:438 devstate_event: device 'DAHDI/1' state '2' -- Executing [s@incoming:1] NoOp(DAHDI/1-1, Incoming s) in new stack [Sep 14 08:21:14] DEBUG[11186]: pbx.c:4058 pbx_substitute_variables_helper_full: Function result is '16' [Sep 14 08:21:14] DEBUG[11186]: pbx.c:4230 pbx_extension_helper: Launching 'Verbose' -- Executing [s@incoming:2] Verbose(DAHDI/1-1, CID 16) in new stack CID 16 [Sep 14 08:21:14] DEBUG[9350]: app_queue.c:1487 handle_statechange: Device 'DAHDI/1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Sep 14 08:21:14] DEBUG[11186]: pbx.c:4230 pbx_extension_helper: Launching 'Set' -- Executing [s@incoming:3] Set(DAHDI/1-1, SPYGROUP=queue-01) in new stack [Sep 14 08:21:14] DEBUG[11186]: pbx.c:4230 pbx_extension_helper: Launching 'Answer' More information available on request. Incidentally, are we entitled to Digium support for these issues on this card? 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] Asterisk HangUp not breaking incoming call for caller
because they're not a member of any queue. 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] Asterisk HangUp not breaking incoming call for caller
not a member of any queue. 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] Asterisk HangUp not breaking incoming call for caller
On Wednesday 12 Sep 2012, Vladimir Mikhelson wrote: Raj, I am just confirming it happens here as well. CentOS 5.7. Asterisk 1.8.15.1. DAHDI 2.6.1. Digium, Inc. Wildcard TDM410 4-port analog card (rev 11) Loadzone = us The problem started manifesting itself after I switched to 1.8.x from 1.6.2.x Typical scenario: a caller apparently hangs up, dial plan goes into voice mail and records a 50 sec message with the CO tones. Then something happens and the line finally gets hung up. I feel this is a different problem: Problem 1: Asterisk receives incoming call from PSTN. Caller hangs up. Asterisk doesn't notice hangup and continues in the dialplan. Problem 2: Asterisk receives incoming call from PSTN. Asterisk eventually executes HangUp(). Caller does not get hung up. Correct me if I'm wrong, but yours seems to be the first, whereas mine is the second. Solved the first one by loading the appropriate zones, loading the Digium card driver with the correct opermode (which must be one of the best-kept secrets on the Internet!) and enabling busydetect in Dahdi. Stuck at the second one for now. 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] can we install 10 PCI card on asterisk
On Monday 27 Aug 2012, DHAVAL INDRODIYA wrote: i would like to know if anyone has done or having idea regarding PRI terminations in asterisk. i have a requirement where i need to support 80 PRI in one machine i have found a machine which have 10 PCI slots available now i am thinking of arranging 8port sangoma card in this pci slots so i can arrenge 10 card in that. Last I checked, the highest channel number DAHDI supported was 1023, limiting you to some 34 E1 PRIs. is it possible to run system like that ? is it good idea , can asterisk handle 2400 calls if machine size and RAM is good. We've faced stability issues with more than 500 simultaneous calls on a single high-powered server, with no transcoding. However, that's probably more a limitation of our own architecture and application than a hard Asterisk limit. 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] can we install 10 PCI card on asterisk
On Tuesday 28 Aug 2012, DHAVAL INDRODIYA wrote: Thanks for everyone input on this, this was just mine thoughts to put 80 PRI line in that.but after reading inputs from everyone i think there are some options to achieve it. it means i need to put a gateway which convert my SIP calls to PRI line and another options is to put multiple asterisk boxes and each box have maximum 16 pri lines . now which is best choice to work on further. also i need to consider hardware sizing too as if gateway is expensive i would go with pri cards. also if i choose gateway then also i need to put multiple asterisk boxes. FWIW... Our largest setup consists of some 2000 SIP users distributed over 2 boxes (sip concentrators). The PSTN interface is through another set of boxes with up to 24 PRIs per box (dial banks). Users log into one of the sip concentrators with soft (Qutecom) or hard SIP phones. When they place a call, it's automatically distributed to a PRI on one of the dial banks. The PRI selection is weighted random, with individual preference sets being assigned for each group of callers. The biggest issue we faced was figuring out that you can't have more than one PSTN provider on a single dial bank -- the timing sources interfere with each other and cause call drops. The current setup connects all the PRIs of a single telco to a single dial bank, eliminating that problem. There are currently 3 telcos providing PRIs in the largest centre. The client and we are happily running vanilla Asterisk Debian packages with (even though I say so myself) some scripting to die for. Setup is completely stable and is being used to generate some $15M of business annually for the client. 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] Check for the voicemail
On Tuesday 21 Aug 2012, Ruben Rögels wrote: Hello, no problem at all, I think this is the tricky part. A smtp dialogue between your email client and a smtp server normally looks like this: user@box:~? netcat mx1.example.com 220 postfix ESMTP mx1.example.com helo me.local 250 mx1.example.com mail from: ruben.roeg...@wiseape.de 250 2.1.0 Ok rcpt to: ruben.roeg...@example.com 450 5.7.1 ruben.roeg...@example.com: Mailbox Full The tricky part is writing or finding a console smtp client that gives you feedback about the 450 error that just happened. Right now I cannot give you a precise way to do that, but I have basic understanding of the technology, so I know that it is possible to do so ;-) I'm looking around in the net, because I think I'll soon have to handle your problem aswell in my company ;-) If I can find solution, I'll post it. Something like this ought to do it: (sleep 5; echo HELO foo; sleep 1; \ echo mail from: f...@example.com; sleep 1; \ echo rcpt to: userid.t...@youwant.to.check; sleep 1; \ echo data; echo test; echo .; sleep 1; echo quit) | \ telnet mail.ho.st 25 21 | fgrep -q '450 5.7.1' notify-user.sh Of course, it's probably better to wrap this into a Perl or equivalent script, but it should work on the shell too. 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] Check for the voicemail
On Wednesday 22 Aug 2012, Roberto Piola wrote: I would simply send the message with sendmail -v and then grep the output for the error message Er, that works too :) Much better solution (as long as you are root). 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] 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
Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
On Friday 03 Aug 2012, C. Savinovich wrote: You don't use 'n's in your dialplan?, you number it yourself? man, what if you have a 300 line dialplan and then you decide to insert a new line in the middle? If you ever used BASIC you'd remember the trick is to increment line numbers (priorities) by 10. I presume a dialplan would work fine even if the priorities aren't sequential, as long as they're increasing monotonically. Could someone confirm? Having said that, I use n with abandon. 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] So long, and thanks for all the fish!
On Tuesday 31 Jul 2012, Kevin P. Fleming wrote: [snip] This is yet another incredibly exciting, career changing opportunity in my life, and I can't wait to see what it will bring. I'll be forever thankful for the opportunity that Digium and the Asterisk community provided me to learn, grow and find the place where my skills and experience are the most valuable (to both myself and my employer). Thanks for all the pertinent and helpful advice and suggestions on the list, Kevin, and all the best in your new assignment. 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] best PRI gateway?
On Sunday 29 Jul 2012, Mike wrote: what are folks using for PRI gateways these days? Obviously there's lots of folks using TE410s and related cards, which work well, and I know reasonably well. However, anyone using anything standalone that stands out as being particularly stellar? One of my clients is using the Redfone TMDoE devices with excellent results. The installation has been in place for over a year. To (try to) answer your specific questions: Anything that: * takes a couple of PRIs or more (or, if it's not costly, I'm not opposed to two of 'em) Takes up to 4 PRIs. * does a good job of echo cancellation (128ms? what's the standard now?) Haven't faced any echo issues so far with up to 400 simultaneous calls on a single box, load balanced over multiple Redfone/PSTN gateways. * isn't horrendously expensive You could get pricing from them. * straightforward to configure Fairly straightforward. The scripts supplied by Redfone work for a single device connected to a computer. We're connecting up to 6 devices (6 x 4 x 30 = 720 lines), and have made our own set of scripts to handle that. * doesn't leave you annoyed every time there's a problem As far as I know my client's blood pressure has remained stable over the non-Redfone = Redfone transition :) 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] CAS T1 - No Ringback
On Friday 27 Jul 2012, Tim Nelson wrote: Another mystery for the list, hopefully someone has ideas on a fix... :) I've got an Asterisk 1.8.12.0 system connected to a CAS T1 (ESF/B8ZS, fractional 1-8). Outbound dialing works correctly, but while the call is in progress, there is no 'ringing' heard by the end user. So, on a SIP phone connected to this system, I dial a number, that call goes out DAHDI via the CAS T1, and the remote side is actually ringing (my cell phone for example), but the SIP phone remains silent. If I answer my cell phone, full 2-way audio is present. Do you Answer() the SIP phone before dialling DAHDI? 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] Finding the position of a character in a string
On Tuesday 24 Jul 2012, Ishfaq Malik wrote:
It there a native asterisk dialplan function which will tell me the
position of a specific character in a given string?
eg if I wanted to find what position the '@' was at in ${SIPURI}
Worst case scenario: write a loop to iterate over each character and
stop at first match.
If you have ODBC installed: your SQL back-end should be able to perform
this function for you.
See if the dialplan function FIELDNUM can be used.
There must be many more ways.
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] chan_sip sending from wrong source address when multiple interfaces are used
On Thursday 12 Jul 2012, Kevin P. Fleming wrote:
On 07/11/2012 11:36 PM, Jeff LaCoursiere wrote:
This does exhibit the problem though. Your OS stack assumes one of
those addresses - the first identified interface? - is the one that
all replies will appear to come from. So phones on the
192.168.2.0/24 network that try to register get replies from
192.168.1.1 and ignore them.
No, I don't think it does. If the server has four interfaces, on
subnets 192.168.{1,2,3,4}.0/24, those are *not* overlapping, and
everything will work as expected. If a UDP packet is received on the
third interface, from an address reachable via routes over that
interface, then the reply to that packet will be sent out over that
same interface, with the source address set to the address assigned
to that interface. Servers are setup this way all the time, and it
works as it should.
Precisely. In fact, if a packet from 192.168.2.n is received on /any/
interface, the response will always go out from the 192.168.2.X
interface. (Barring some weird routing/iptables configuration, of
course.)
There must be more to the network configuration than something this
simple in order to cause the IP stack on the Asterisk server to
choose the wrong source IP address for outbound packets.
I've usually seen this (wrong interface chosen for outbound) happen when
NAT is in the picture. However, the OP doesn't mention any NAT-related
configuration.
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] PSTN termination in Virtualized Asterisk Environment
On Thursday 31 May 2012, David Klaverstyn wrote: I would not recommend the redfone devices. I have had bad experiences with them. Have a look at the Digium G100 or G200 devices. They look far superior. Hmm, that's odd, since we've had an excellent experience with Redfones (purchased through Mitul, incidentally) overall. Our client is running Asterisk servers doing 1/2 a million outbound + some inbound through Redfone devices every day with no significant issues. This is spread over some 4 or 5 PSTN providers country-wide. Overall I'd say our experience with the Redfones has been very positive, leaving aside the advantage of having an E1 interface outside of the computer (device and computer can be power cycled independently if required, device can be easily interfaced to a different computer, etc.) 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] Extensions routing
On Saturday 19 May 2012, Mikhail Lischuk wrote: I've been playing around with clustering some Asterisk servers for sake of fail-over and load balancing with DNS round-robin, and came to one problem. If I have, say, 2 servers, and clients register either on 1 or 2, how can I route extensions between them? I mean, if today user with extension 101 is registered on server1, and tomorrow he will register with server2 - how would any of servers know where to route it? Won't Dundi serve your purpose? From http://www.dundi.com/ : DUNDi™ is a peer-to-peer system for locating Internet gateways to telephony services. Unlike traditional centralized services (such as the remarkably simple and concise ENUM standard), DUNDi is fully-distributed with no centralized authority whatsoever. DUNDi is not itself a Voice-over IP signaling or media protocol. Instead, it publishes routes which are in turn accessed via industry standard protocols such as IAX™, SIP and H.323. DUNDi can be used within an enterprise to create a fully-federated PBX with no central point of failure, and the ability to arbitrarily add new extensions, gateways and other resources to a trusted web of communication servers, where any adds, moves, changes, failures or new routes are automatically absorbed within the cloud with no additional configuration. 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] Least Machine Specs to run a production asterisk server
On Friday 11 May 2012, Carlos Alvarez wrote: On Fri, May 11, 2012 at 6:15 AM, eherr email.eherr9...@gmail.com wrote: What is the lowest end machine to run a production asterisk server. Depends on a lot of variables. I've got some old 1.8GHz 1U servers running hundreds of calls. How many calls, how much transcoding, etc etc. You can run Asterisk on a Linksys WRT router, so I guess that's about a minimum. Don't know where this fits into the minimal requirements, but there's also a working (alpha) Iphone port: http://mgamble.ca/oss/iphone_asterisk/ Probably more destined to be a novelty than anything else. 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] Account code script needed.
On Tuesday 17 Apr 2012, cjwstudios wrote: Looking for quotes on a very simple script that will require a pin number before allowing a call to be placed. The pin number would be recorded to their mysql CDR. Thank you. Will the DISA application do what you need? 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] Problem installing asterisk 10.1.3 on SUSE
On Thursday 22 Mar 2012, Danny Nicholas wrote: Hi gang, I've put 10.X on about 15 different VM's now, but I've run into a buzzsaw on this one and my google-fu has failed me Output of make CC=cc CXX= LD= AR= RANLIB= CFLAGS= make -C menuselect CONFIGURE_SILENT=--silent menuselect make[1]: Entering directory `/usr/local/src/asterisk-10.1.3/menuselect' make[1]: `menuselect' is up to date. make[1]: Leaving directory `/usr/local/src/asterisk-10.1.3/menuselect' [LD] astdb2sqlite3.o db1-ast/libdb1.a - astdb2sqlite3 /usr/lib64/gcc/x86_64-suse-linux/4.1.2/libstdc++.so: file not recognized: File format not recognized Try: file astdb2sqlite3.o file db1-ast/*.o file /usr/lib64/gcc/x86_64-suse-linux/4.1.2/libstdc++.so (with the appropriate paths for the first two) and see if they're the same (32 or 64-bit) architecture. The third is likely to be 64-bit anyway, what are the first two? 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] Rate sheet normalization
On Thursday 15 Mar 2012, Markus wrote: With like 10 different ratesheets from 10 different providers, of which many change their rates every few days, manually doing it in Excel is too time consuming... Is it possible to get samples? I'd be interested in looking into developing a script that can handle this problem generically, and presumably you're available to alpha- and beta-test in any case :) 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] Normal ringing tone for the caller, while call waiting.
On Wednesday 14 Mar 2012, NaJIm wrote: When I make a call to an extension, which is on another call, the called party (who is on call waiting) will get a BEEP sound. But the caller gets the normal ringing tone. Is there any way to have a different dialer tone for the Caller, which lets him know that the other person is on a call.. i.e. When A calls B, while B is already on a call with C, Is there a way to let A get a message that B is busy on another call. If SIP, have a look at the busylevel and call-limit sip.conf parameters. 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] Capacity of single instance of Asterisk
On Tuesday 13 Mar 2012, Amit Patkar | Avhan Technologies Pvt Ltd wrote: Thank for your views. Where as no one is ready to share real numbers. I am looking at benchmarks so that I can plan for resources. Since asterisk project is active for so many years, I was expecting some published numbers. We're running some 400 simultaneous calls with recording and no transcoding on a 2xQuad-core Intel boxes, 16GB RAM. The box is serving SIP clients and passes calls over an IAX2 trunk to the PSTN-connected box. Load average rarely goes above 0.5. Recording is done on a RAID array attached to a separate SCSI controller, which makes a lot of difference to performance. 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] Group write permissions /etc/asterisk/.
On Tuesday 06 Mar 2012, Jason Parker wrote: I don't know if I would call it a bug since the switch to install was intentional, but I wouldn't say it's necessarily expected either. I don't really have a strong opinion either way though. If anything, I might be inclined to argue that 750 (or 770) would be more appropriate. Considering that (e.g.) sip.conf and iax.conf may contain passwords in clear-text, I'd agree that 770/750 for directories and 660/640 for files would be most appropriate. The g+w bit needs to be set only on those directories/files that ought to be writable from within the Asterisk process itself. 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] Forwarding queue to remote agent over PSTN
Hi, A client is looking for a way to have queue agents available over their mobile or land-line phones. In other words, some queue members would be local (over SIP channels) while others would only be reachable by dialling their (mobile) phones over the PSTN. Is there some easy way to accomplish this? 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] Forwarding queue to remote agent over PSTN
On Thursday 16 Feb 2012, Satish Barot wrote: If you have your agents static(hard coded) for Queue in queues.conf, you could add following in your queue definition, member = DAHDI/G0/XX Replace XX with your Agent's cellphone or Landline number. And if you have your Agents added dynamically in Queue, use local channel as a Queue member and have your local channel dial the cellphone or Landline number. See the 'Using Local Channels' section on a link http://ofps.oreilly.com/titles/9780596517342/asterisk-ACD.html for more information. (Courtesy:Leif Madsen, Jim Van Meggelen, and Russell Bryant) That's brilliant, and answers all my questions. Thanks! Also thanks to Danny Nicholas for the initial push in the right direction. Regards, -- Raj On Wed, Feb 15, 2012 at 6:32 PM, Raj Mathur (राज माथुर) r...@linux-delhi.org wrote: A client is looking for a way to have queue agents available over their mobile or land-line phones. In other words, some queue members would be local (over SIP channels) while others would only be reachable by dialling their (mobile) phones over the PSTN. Is there some easy way to accomplish this? -- 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] Binding to 0.0.0.0 a security risk?
On Tuesday 07 Feb 2012, Jakob Hirsch wrote: Steve Edwards, 2012-02-06 01:43: Unfortunately, (IIRC) Asterisk does not reply to the same interface packets are received from which limits the usefulness of multiple interfaces. Right, that's what I also observed. We had to take special measures to handle this. The problem lies in the nature of connectionless protocols as UDP. We also use freeradius, which does it right by itself (but still needs a compile time switch --with-udpfromto for it). Packets not going out on the same interface as the one they were received on is a general IP issue, not just for connectionless protocols. The same behaviour can be seen with TCP too. Unless you mangle with iptables or something, all information about the received interface has been stripped from the packet by the time it reaches the IP layer. /nitpick 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] Binding to 0.0.0.0 a security risk?
On Tuesday 07 Feb 2012, Josh wrote: [snip] Unfortunately, (IIRC) Asterisk does not reply to the same interface packets are received from which limits the usefulness of multiple interfaces. What do you mean by that? If a request is received over eht1 are you saying that Asterisk does not respond over the same interface?! As far as I know, Asterisk would use the default Linux/Unix routing algorithms to send packets out, in which case yes: responses may not go out on the same interface packets were received on. E.g. if you receive packets with non-LAN IP addresses on eth0, while your default route is set to eth1, in the absence of custom routing Linux will send the responses over eth1. There are ways to overcome this for specific situations, but no general method that I'm aware of (though I'm happy to be corrected). 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] Script to automatically update externip. Useful for a host with dynamic public IP
On Monday 06 Feb 2012, John Cahill wrote: logger -s checksetexternip.sh: External IP address has changed, changing /etc/asterisk/sip_general_custom.conf grep -v externip /etc/asterisk/sip_general_custom.conf /etc/asterisk/sip_general_custom.conf.tmp echo externip=$EXTERNIP /etc/asterisk/sip_general_custom.conf.tmp cp /etc/asterisk/sip_general_custom.conf.tmp /etc/asterisk/sip_general_custom.conf rm /etc/asterisk/sip_general_custom.conf.tmp You could also do something like: sed -i -e s/^externip *=.*/externip = $EXTERNIP/ /etc/asterisk/sip.conf Apologies for the wrapped code. 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] Queue member is permanently BUSY
Hi, we have a queue with some 20 agents. Agents are defined statically in queues.conf (in an include, actually). One of the agents is showing up as NOT AVAILABLE in queue show... when his client is disconnected from Asterisk. However, the moment his system gets connected, his status in queue show... changes to BUSY, even though he's not taking any calls. We've tried both soft and hard phones as clients for this agent, but the problem remains the same. Any help appreciated. 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] best softphone for 2012?
On Saturday 07 Jan 2012, Tom Poe wrote: Just installed asterisknow 1.6. I can access freepbx. I need to test system on my LAN. Which softphone is best to use? I'm running ubuntu on Dell optiplex G260 desktop at home. I'm hoping to setup basic IP PBX for incoming/outgoing calls. No video. We've tried out Qutecom, Linphone and SFLPhone, and all three work well for our client's environment (call-out centre). We finally implemented Qutecom for most of the desktops, for some reason that escapes my memory right now. Ekiga was just too heavy for our needs. There's also Twinkle, which is what we use for testing since it's light and fast. 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] Mark queue agent as away
Hi, I have a queue with a number of (static) agents. Is there an easy way for an agent to indicate that she is away from her seat, so that her phone is not rung when a call comes in? And the converse, of course: be able to notify Asterisk when she is back and ready to accept calls? 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] Asterisk 1.8 warns for lines starting with # in /etc/dahdi/system.conf
On Thursday 22 Dec 2011, Olivier wrote: Testing 1.8.8.0 (with Dahdi 2.5.0.2 and asterisk-gui 2.1.0-rc1), I'm seeing this on my console: WARNING[25363]: config.c:1208 process_text_line: Unknown directive '#' at line 1 of /etc/asterisk/../dahdi/system.conf This warning is repeated for every line starting with a # char. Shall I care ? Suggestions ? (To me, /etc/asterisk/../dahdi/system.conf belongs to Dahdi, not to Asterisk) Use ; instead of # to comment dahdi/system.conf. If it's auto- generated, perhaps this would help after the auto-generation: sed -i -e 's/^#/;/' /etc/dahdi/system.conf 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] India Telecom regulations
On Tuesday 20 Dec 2011, Steve Edwards wrote: On Mon, 19 Dec 2011, Nick Khamis wrote: SIP in India is illegal. What about IAX, Skype, VPN, etc? The only thing that is not permitted is bridging Internet calls with the Indian PSTN. In fact, that too is allowed if you have a VoIP licence from the government. Apart from that, as long as you continue using it within your own organisation, any protocol is fine. IANAL. TINLA. 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] India Telecom regulations
On Tuesday 20 Dec 2011, khalid touati wrote: Thank you Raj, so with VOIP license calls can go beyond our pbx to PSTN (india), right, if so this what i needed to know to call Indian cellphone from US (or other countries) If your objective is to originate calls in the US (using whatever technology), route them over SIP and then terminate them to the PSTN in India, then yes: your Indian presence would need a VoIP licence. Similarly for the reverse: originate a call from Indian PSTN to your local office here and route it using VoIP to any destination (whether within India or abroad). A licence is required in that case too. In general, interconnection of two different entities by bridging Indian PSTN with any other technology requires a licence. If you're only doing VoIP-VoIP, or PSTN-PSTN, or bridging an Indian VoIP call to PSTN outside India then it's permitted in principle. This is why, e.g., Skype is permitted: it doesn't connect to the Indian PSTN at any stage. Once again, IANAL and TINLA. This is purely from my (mostly informed) understanding of the current laws. 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] How to see initiall dialled extension in CDR records ?
Please start a new thread for new conversations.
On Monday 12 Dec 2011, Albert wrote:
I have following problem. For statistical reasons I need to know what
was initiall number dialled by customer. I have 2 premium numbers,
for which customers are billed differently per minute. But in my CDR
table i can see only last dialled extension from voice menu. In this
example it shows me that customer (077XXX) was billed 293
seconds but i am not seeing which number he dialed. Only what i can
see is last destination he choosed from voice menu which was '1'.
What shall I do to see which number was dialled initially?
ps. Original numer was X-ed.
calldate |clid|src | dst |
dcontext |channel | dstchannel | lastapp
| lastdata | duration | billsec |
| disposition
|
| amaflags | accountcode | uniqueid| userfield
-+++-+---
-++++
+--+-+-+-
-+-+---+---
2011-12-12 15:11:26 | 077XXX | 077XXX | 1 |
stories-en-options | DAHDI/i1/077XXX-13 || Playback
| custom/l_stories/en/en_bible | 293 | 293 | ANSWERED|
3 | | 1323691886.91 |
Below is part on my extensioans file and voice menu structure
[incoming-calls-from-e1-span1]
exten = 902000111,1,Verbose(Call is comming ${EXTEN})
exten = 902000111,n,Goto(voiceservices_menu,2666,1)
exten = 902000222,1,Verbose(Call is comming ${EXTEN})
exten =
902000222,n,GotoIfTime(08:00-22:59,mon-sun,*,*?supportmenu,2555,1)
exten = 902000222,n,Playback(custom/l_line/callcenter-closed) exten
= 902000222,n,Goto(voiceservices_menu,2666,1)
If I understand your problem correctly, you have a few options:
1. Answer() the call in the 902000XXX extension. That will cause the
CDR to be written with that extension as the dst.
2. Save the called number in the CDR userfield in the 902000XXX
extension itself using something like:
exten = 902000111,n,Set(CDR(userfield)=${EXTEN})
There're probably many other (better) ways of achieving this too.
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] Automatic IVR generator (alpha)
Hi, Have started work on a Perl script to automatically generate Asterisk IVR dialplans from a YAML configuration. It's pretty rudimentary right now, but working for the couple of test cases I've thrown at it. Current features: - Built-in navigation (GoTop, GoToMenu). A GoUp is planned. - Default 'i' stanzas for menus that don't define one. - Any level of menu nesting. - Special pre-entry menu (for auth, language selection, etc.). - Priority labels. The Perl script and the test YAML file are respectively available from: http://etc.kandalaya.org/IVR/ivr.pl http://etc.kandalaya.org/IVR/test.yml First off, does this look like a useful tool? And if it does, what features would you like to see in the finished product? (No promises about speedy implementation, but I'll try!) 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] SIP registration issues
Hi, Having problems with a client trying to login to Asterisk 1.6.2 from behind a DSL router. The account can be accessed perfectly from other clients. Would appreciate if you could look at the the attached log and see if you spot any glaring issues. The user is very infrequently available for discussion and testing, so please try to batch questions in one mail itself! 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 REGISTER sip:SERVER-IP SIP/2.0 CSeq: 100 REGISTER Via: SIP/2.0/UDP CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;rport User-Agent: Ekiga/3.2.7 From: sip:ACCOUNT-ID@SERVER-IP;tag=0d5779ba-a505-1910-8afe-001302871a3d Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway To: sip:ACCOUNT-ID@SERVER-IP Contact: sip:ACCOUNT-ID@CLIENT-IP:49153;q=1, sip:ACCOUNT-ID@CLIENT-IP:49152;q=0.667, sip:ACCOUNT-ID@CLIENT-IP:49152;q=0.334 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Expires: 3600 Content-Length: 0 Max-Forwards: 70 - --- (12 headers 0 lines) --- Sending to CLIENT-IP : 49153 (no NAT) --- Transmitting (no NAT) to CLIENT-IP:49153 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;received=CLIENT-IP;rport=49152 From: sip:ACCOUNT-ID@SERVER-IP;tag=0d5779ba-a505-1910-8afe-001302871a3d To: sip:ACCOUNT-ID@SERVER-IP;tag=as5d35c321 Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway CSeq: 100 REGISTER Server: Asterisk PBX 1.6.2.9-2+squeeze3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=09c83283 Content-Length: 0 Scheduling destruction of SIP dialog '0d5779ba-a505-1910-8afd-001302871a3d@Gateway' in 32000 ms (Method: REGISTER) --- SIP read from UDP:CLIENT-IP:49152 --- REGISTER sip:SERVER-IP SIP/2.0 CSeq: 100 REGISTER Via: SIP/2.0/UDP CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;rport User-Agent: Ekiga/3.2.7 From: sip:ACCOUNT-ID@SERVER-IP;tag=0d5779ba-a505-1910-8afe-001302871a3d Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway To: sip:ACCOUNT-ID@SERVER-IP Contact: sip:ACCOUNT-ID@CLIENT-IP:49153;q=1, sip:ACCOUNT-ID@CLIENT-IP:49152;q=0.667, sip:ACCOUNT-ID@CLIENT-IP:49152;q=0.334 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Expires: 3600 Content-Length: 0 Max-Forwards: 70 - --- (12 headers 0 lines) --- Sending to CLIENT-IP : 49153 (no NAT) --- Transmitting (no NAT) to CLIENT-IP:49153 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;received=CLIENT-IP;rport=49152 From: sip:ACCOUNT-ID@SERVER-IP;tag=0d5779ba-a505-1910-8afe-001302871a3d To: sip:ACCOUNT-ID@SERVER-IP;tag=as5d35c321 Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway CSeq: 100 REGISTER Server: Asterisk PBX 1.6.2.9-2+squeeze3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=09c83283 Content-Length: 0 Scheduling destruction of SIP dialog '0d5779ba-a505-1910-8afd-001302871a3d@Gateway' in 32000 ms (Method: REGISTER) --- SIP read from UDP:CLIENT-IP:49152 --- REGISTER sip:SERVER-IP SIP/2.0 CSeq: 100 REGISTER Via: SIP/2.0/UDP CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;rport User-Agent: Ekiga/3.2.7 From: sip:ACCOUNT-ID@SERVER-IP;tag=0d5779ba-a505-1910-8afe-001302871a3d Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway To: sip:ACCOUNT-ID@SERVER-IP Contact: sip:ACCOUNT-ID@CLIENT-IP:49153;q=1, sip:ACCOUNT-ID@CLIENT-IP:49152;q=0.667, sip:ACCOUNT-ID@CLIENT-IP:49152;q=0.334 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Expires: 3600 Content-Length: 0 Max-Forwards: 70 - --- (12 headers 0 lines) --- Sending to CLIENT-IP : 49153 (no NAT) --- Transmitting (no NAT) to CLIENT-IP:49153 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP CLIENT-IP:49153;branch=z9hG4bK072ae5cc-a505-1910-8b01-001302871a3d;received=CLIENT-IP;rport=49152 From: sip:ACCOUNT-ID@SERVER-IP;tag=0d5779ba-a505-1910-8afe-001302871a3d To: sip:ACCOUNT-ID@SERVER-IP;tag=as5d35c321 Call-ID: 0d5779ba-a505-1910-8afd-001302871a3d@Gateway CSeq: 100 REGISTER Server: Asterisk PBX 1.6.2.9-2+squeeze3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=09c83283 Content-Length: 0 Scheduling destruction of SIP dialog '0d5779ba-a505-1910-8afd-001302871a3d@Gateway' in 32000 ms (Method: REGISTER) --- SIP read from UDP:CLIENT-IP:49152 --- REGISTER sip:SERVER-IP SIP/2.0 CSeq: 100 REGISTER Via: SIP/2.0/UDP CLIENT
Re: [asterisk-users] Licensing question.
On Wednesday 09 Nov 2011, Yaroslav Panych wrote: I shall contact when(and if) decision will be made. But such decision cannot be made basing only on this paragraph, because it does not describes anything. There are no description of licensing procedure, nor pricing, nor liability, rights or freedoms(at least in general approximation) of sides. So I'm here asking and asking again. In any case, even usage of GPL-ed copy of Asterisk(or any other software) is illegal in my country. If I understand your situation correctly, the solution is very simple and two-fold: 1. You want to develop and distribute FOSS (Free and Open-Source Software) extensions for Asterisk. These extensions may be modules or enhancements. In this situation, just go ahead and develop the extensions and distribute them under GPLv2. 2. You want to develop and distribute extensions for Asterisk but don't want to release their source. In this situation, get a source code licence from Digium (or whoever has Asterisk source copyright) under a non-GPLv2 licence. Develop and release your extensions under a proprietary licence. You have no obligation to release their source, since your extensions are derived from a non-GPL licence. Of course, Digium (or whoever) will impose conditions on your use of the source they license to you, but that is between you and Digium (or whoever) and doesn't concern anyone else. This is obviously very generalised and doesn't cover all cases, but it does cover the basic dual-licensing policy. I am not a lawyer. This is not legal advice. In fact, as others have also stated, it is highly recommended you consult with a lawyer who is well-versed in licensing and specially FOSS licensing. 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] [OT] Re: Licensing question.
On Wednesday 09 Nov 2011, Kevin P. Fleming wrote: [snip] * The GPLv2 places no restrictions on what you can 'write', it only places restrictions on your distribution of things that you write that could be considered 'derivative works' of a GPLv2-covered work (in this case, Asterisk). If you write something that could be considered a derivative work, and you wish to distribute it, then the GPLv2 obligates you to distribute that work under the GPLv2 or a compatible license. Minor nitpick: a derivative of a GPLv2 work can only be released under the GPLv2, or a licence so similar to GPLv2 as to be indistinguishable from it. You cannot distribute a GPLv2 derived work under, e.g. a BSD or Artistic licence. 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] Starting asterisk turns bash console text white in rxvt
On Monday 31 Oct 2011, Sebastian Arcus wrote: Every time I start Asterisk (just by issuing /usr/sbin/asterisk), the bash console text turns white. I'm using rxvt, so this makes everything pretty much invisible. If I login into the Asterisk console (asterisk -rvvv) - the text turns black again. This is not critical, but quite annoying. I've experienced it both with 1.6 and 1.8 installations. Maybe, just maybe this is a problem with rxvt - but I use it for absolutely everything on the command line - and no other application has problems of this type. Try adding -n to the asterisk command-line arguments. That is supposed to turn off colours (though I've had mixed results with it). Also see the -B and -W options (again, not always effective for some reason). Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- 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] Queue announcements and MOH blanking out on calls from PSTN over IAX2
On Monday 31 Oct 2011, Alex Kauffmann wrote: Sorry if i missed it, but is IAX2 trunked? IF so, perhaps you are running out of bandwidth in your IAX2 trunk. The setting 'trunkmaxsize' defaults to 128000 bytes. From the readme file: ...Once this limit is ; reached, calls may be dropped or begin to lose audio. Depending on the codec in use and ; number of channels to be supported this value may need to be raised, but in most cases the ; default value is large enough. Normal calling (in and out) continues to work fine while this problem is being seen. We're not trunked, though -- each call sets up and tears down the connection. Just finished some preliminary testing on the same Asterisk on a 32-bit (instead of 64-bit) machine and the problem is not appearing. I'm reluctant to stamp this as the solution right now, since the test machine doesn't have any SIP peers at all (that's the only difference in the production and test beds), but am still hopeful. An i686 Asterisk installation on the production server should confirm or deny the theory that the problem was purely for Asterisk x64. Will post once we know more. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- 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] Queue announcements and MOH blanking out on calls from PSTN over IAX2
On Sunday 30 Oct 2011, Raj Mathur (राज माथुर) wrote: After looking further, the problem seems to be purely in playing recorded messages over IAX2. Looking at the debug logs on the SIP server (which is playing the recorded messages) shows that it stops playing one of the messages at some point in the flow, and then never plays anything again. This seems to be very similar to: https://issues.asterisk.org/view.php?id=17232 except there is no virtualisation involved in the process -- everything is working on native hardware. It /is/ amd64 Debian Squeeze running on Intel, though. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- 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] Queue announcements and MOH blanking out on calls from PSTN over IAX2
On Sunday 30 Oct 2011, Sammy Govind wrote: hmmm so IAX channel is playing with you guys. 1- Cant you guys use SIP, does this happen with SIP trunk as well !? 2- Which version of asterisk are there on both servers. 3- See the output of the command core show file versions in your both asterisk servers. Mainly lookout for IAX channel version. Also try enabling IAX debug and paste the output on console. 1.6.2.9-2+squeeze3 on the SIP server, 1.6.2.9-2+squeeze1 on the Dial server. I doubt if we'll be able to change the architecture of an infrastructure handling up to 450 simultaneous calls for the past 6 months at this stage, so SIP is out. IAX2 has been working beautifully for our needs up to this point, and we need to find a solution that we can integrate into this architecture itself! Incidentally, if anyone's interested, the installation itself is detailed at: http://www.mail-archive.com/ilugd@lists.linux-delhi.org/msg28166.html Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- 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] Reporting for Asterisk Call Center
On Sunday 30 Oct 2011, bilal ghayyad wrote: In case I need to retreive the real time data (for example, how many calls currently in the queue, and how many calls currently waiting in the queue, how many agents currently are logged in ... etc). How to get this? Is it using the AGI? From where I can get information about this? Because in the CDR, there is nothing mention or can be obtained for these informations (how many in the queue and how many is waiting .. etc), correct? On the command-line you can give: queue show queue-name which will give you real-time information about the queue. Presumably you can do the same through AGI too. In addition, I believe there are some ready-made packages (both FOSS and proprietary) that will display this information nicely formatted. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- 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] r...@linux-delhi.org
On Sunday 30 Oct 2011, bilal ghayyad wrote: Actually I need to do a dash board for reporting, so I beleive the only way is to use the AGI, correct? But where I can find documents or link that can help me to do this? About ur sentence: some ready-made packages (both FOSS and proprietary) that will display this information nicely formatted. What is the FOSS and proprietary? Any link for it? And this ready-made packages can work with asterisk 1.8? FOSS is Free and Open Source Software, like Asterisk and Linux; Proprietary is software like Windows, which you cannot distribute and modify. For Queue statistics, http://www.asternic.biz/ has both FOSS and proprietary versions of its package. http://queue-tip.rubyforge.org/ is FOSS. http://sourceforge.net/projects/astacd-activity/ is FOSS. Disclaimer: I haven't used any of these packages, and there must be many more that my quick search didn't find. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- 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] Queue announcements and MOH blanking out on calls from PSTN over IAX2
On Saturday 29 Oct 2011, Raj Mathur (राज माथुर) wrote: [snip] Callers coming in from the PSTN (through the Dial server, over IAX2) can also talk normally after an agent has picked up the call. However, callers from the PSTN get the announcement and/or MOH blanked out after a random period of time, typically 5-10 seconds. This often happens in the middle of the queue position or thank-you announcement. After the blanking out, the call is still alive, queue functions are working, and if an agent picks up the calls s/he can talk normally to the caller. However, blanking out of the MOH/announcement makes the caller think that the call has been dropped, and they hang up before an agent answers. Debug logs show that Asterisk is playing the MOH and announcement files continuously even though the caller cannot hear them. Unable to figure out why the blanking happens ONLY on incoming calls from the PSTN. Any help appreciated. Further simplified the issue to an extension that just does: ... Answer() ... MusicOnHold(default) When called from the PSTN, the musiconhold blanks out after a few seconds, while it plays fine when the extension is called locally. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- 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] Queue announcements and MOH blanking out on calls from PSTN over IAX2
-2655, 1) in new stack -- Executing [6000@cg-20:32] Queue(IAX2/dialbank-1-2655, cg-20,dr) in new stack -- IAX2/dialbank-1-2655 Playing 'queue-youarenext.slin' (language 'en') -- Told IAX2/dialbank-1-2655 in cg-20 their queue position (which was 1) -- IAX2/dialbank-1-2655 Playing 'queue-thankyou.slin' (language 'en') -- IAX2/dialbank-1-2655 Playing 'queue-youarenext.slin' (language 'en') *** Caller hung up after some time of silence == Spawn extension (cg-20, 6000, 32) exited non-zero on 'IAX2/dialbank-1-2655' -- Hungup 'IAX2/dialbank-1-2655' All this works perfectly when tested from a client logged in to the SIP server itself. The problem only arises when a caller calls the Dial server from the PSTN, which then passes the call to the SIP server over IAX2. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- 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] Queue announcements and MOH blanking out on calls from PSTN over IAX2
Hi, Problem with Asterisk 1.6.2.9 on Debian Squeeze. * Infrastructure We have two servers, SIP and Dial. The SIP server handles SIP clients; it also receives incoming PSTN calls from the Dial server and makes outgoing PSTN calls on the Dial server. The Dial server is connected to multiple 4-port Redfone devices for handling PSTN incoming and outgoing calls. Outgoing calls always originate from and incoming calls always terminate at the SIP server. SIP and Dial servers are connected over IAX2. Normal incoming and outgoing have been working fine for many months now on this setup. * Problem We recently enabled caller queues on the SIP server. Queue functions are working fine. Local (on SIP server itself) callers get periodic position announcements and MOH while they wait for the call to be picked up. Once an agent picks up the call, the caller and agent can talk normally. Tried with an IAX2 client (instead of SIP) and that works fine too. Callers coming in from the PSTN (through the Dial server, over IAX2) can also talk normally after an agent has picked up the call. However, callers from the PSTN get the announcement and/or MOH blanked out after a random period of time, typically 5-10 seconds. This often happens in the middle of the queue position or thank-you announcement. After the blanking out, the call is still alive, queue functions are working, and if an agent picks up the calls s/he can talk normally to the caller. However, blanking out of the MOH/announcement makes the caller think that the call has been dropped, and they hang up before an agent answers. Debug logs show that Asterisk is playing the MOH and announcement files continuously even though the caller cannot hear them. Unable to figure out why the blanking happens ONLY on incoming calls from the PSTN. Any help appreciated. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- 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] Dialplan help: hang up incoming call and call the number back
On Tue, Mar 29, 2011 at 12:23 PM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: design your dial-plan for routing a specific number to different context , you can try func_odbc for query to DB if you have a large number of setup. ideally its called click to call but you are made it as, miss call and you will get a call. On Mon, Mar 28, 2011 at 5:21 PM, Roger Burton West ro...@firedrake.org wrote: On Mon, Mar 28, 2011 at 05:14:50PM +0530, Raj Mathur wrote: Is there a better way of handling the post-hangup processing? Callfiles? Thanks, call files worked beautifully. Takes a couple of commands to make the call file in Asterisk (I didn't want to call any heavy external programs like Perl or Awk, though that would have been more elegant), and the rest works out of the box. Regards, -- Raj -- _ -- 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] Discover when remote phone answers through IAX2
Hi, I'm using IAX2 between our SIP and PSTN servers, both running Asterisk 1.6.2. Users connect to the SIP server and dial; the SIP server forwards the call to the PSTN server over IAX2, which then dials out over the connected PRI. Since users need detailed call progress feedback, the first action in the dialplan on the PSTN server side is Answer(). In this scenario it's easy for a human to know when a call has been answered. However, the SIP-side Asterisk treats the call as answered the moment the PSTN server executes Answer(). Is there any way of determining on the SIP side when the called party actually picks up the phone? Or if she doesn't, the status of the call as it progresses? Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- 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] Dialplan help: hang up incoming call and call the number back
Hi, I'm trying to setup Asterisk so that: 1. I call a specific number that goes to a defined extension from my phone (an external line). 2. Asterisk notes my phone number (the CLID) and hangs up without picking up the call. 3. Asterisk initiates a call to my phone and prompts me for a passkey. 4. Asterisk validates the passkey and lets me enter another number (say FOO). 5. Asterisk dials FOO on my behalf and lets me talk to FOO. I'm currently using extension h for handling the post-hangup processing; however this seems to involve a lot of validation (since other calls are also terminating into the same context and, eventually, getting hungup) and has issues passing variables between extensions. Is there a better way of handling the post-hangup processing? Regards, -- Raj -- _ -- 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] dahdi restart warning
On Friday 25 Mar 2011, satish patel wrote: *CLI*CLI dahdi restart [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'userbase' (on reload) at line 23. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'vmsecret' (on reload) at line 31. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'hassip' (on reload) at line 35. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'hasiax' (on reload) at line 39. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'hasmanager' (on reload) at line 47. [Mar 23 14:01:10] WARNING[4315]: sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel anyway! I've seen the last error (No D-channels available) on multiport devices where some of the ports are not populated. So if you have a 4-port PRI device with just one PRI plugged in you would get that error regularly on the remaining 3 PRI ports. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- 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] [Opinion Request] SIP phones that work well with Asterisk
Hi, Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. Thanks, -- Raj -- _ -- 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] [Opinion Request] SIP phones that work well with Asterisk
On Wednesday 09 Mar 2011, Raj Mathur wrote: Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. Thanks to all who replied. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- 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] ChanSpy with alphanumeric SIP channels [1.6.2]
Hi, I'm using SIP users of the form 'ab_12345' (two letters, underscore, 5 digits). ChanSpy is working fine for listening in to conversations initiated by these channels, and I can use '*' to randomly switch channels. However, is there any way in this scenario to be able to switch ChanSpy to a specific channel by typing in a ...# key sequence during a spy session? Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- 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] [NEWBIE] Simple hunt group on SIP -- need confirmation
Hi, Is doing this as simple as just creating a queue in queue.conf? I have the following setup: 1. Single hunting 1-800 number mapped to multiple numbers in a hunt group by the telco. 2. All calls land up at Asterisk over SIP. Now I need: 3. A hunt group of operator extensions mapped to the incoming lines. 4. Music-on-hold if all the operator extensions are busy. Queues seem to handle this trivially, but I'm not sure if there are any gotchas. Your advice would be appreciated. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- 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
