Re: [asterisk-users] How to monitor non-SNMP SIP devices ?
Quoting Ishfaq Malik (i...@pack-net.co.uk): On 9 July 2014 16:19, Olivier oza.4...@gmail.com wrote: Hi, I'm seeing a trend in which SIP devices such as Yealink SIP phones (with v72 firmware), are dropping support of SNMP in favor of HTTP eventing How do deal with those devices ? If you set qualify on your peers you could monitor the event stream of the AMI which would show you any end point going unreachable. This is what i do. Certain 'important' SIP endpoints have a qualify setting in Asterisk and i use AMI (or 'asterisk -rx ...') to query that state with an SNMP-extend hook. HTH. -Sndr. -- | What do sheep count when they want to fall asleep? | 4096R/20CC6CD2 - 6D40 1A20 B9AA 87D4 84C7 FBD6 F3A9 9442 20CC 6CD2 -- _ -- 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] 'restart when convenient'
Hi, I want to do a scripted 'restart when convenient' on a daily basis. This used to work, but since i've upgraded to Asterisk 11.7 it seems it's never convenient to restart the server. My question: how can i tell *why* it's not convenient to restart the server? It used to be some colleague left the receiver OffHook or something like that, but even when i'm fairly certain there's no activity on my pbx, all receivers are OnHook, Asterisk won't restart. How can i debug why Asterisk thinks it isn't convenient to restart? Thanks! -Sander. -- | Hey.. I'm done talkin'. Now check out my pretty! | 4096R/20CC6CD2 - 6D40 1A20 B9AA 87D4 84C7 FBD6 F3A9 9442 20CC 6CD2 -- _ -- 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] 'restart when convenient'
Quoting jonathan white (j...@uvacity.com): I'm doing a scripted restart by using the asterisk command line to tell me how many active calls are current. If 0 then restart. For this you use 'core show calls'? If 0, do you 'core restart now' or do you kill and restart Asterisk in any other way? -- | If you're going to be weird, be confident about it. | 4096R/20CC6CD2 - 6D40 1A20 B9AA 87D4 84C7 FBD6 F3A9 9442 20CC 6CD2 -- _ -- 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] 'restart when convenient'
Quoting D'Arcy J.M. Cain (da...@vex.net): OP - can you clarify what actual command you are running? pbx1*CLI core [?] abortclearping reload restart set show stop waitfullybooted pbx1*CLI core restart [?] gracefully now when pbx1*CLI core restart when [?] pbx1*CLI core restart when convenient I use 'core restart when convenient'. This is Asterisk 11.7.0~dfsg-1ubuntu1. -- | Zebras are colored with light stripes on a dark background. | 4096R/20CC6CD2 - 6D40 1A20 B9AA 87D4 84C7 FBD6 F3A9 9442 20CC 6CD2 -- _ -- 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] 'restart when convenient'
Quoting D'Arcy J.M. Cain (da...@vex.net): OP - can you clarify what actual command you are running? I use 'core restart when convenient'. Right. You said restart when convenient in your original email. Sorry for the confusion. But the problem is not what command i am using, the problem is Asterisk not restarting and it's obscure to me why it is not restarting. pbx1*CLI core show channels Channel Location State Application(Data) 0 active channels 0 active calls 1925 calls processed pbx1*CLI sccp show devices [ no devices show 'Act'ivty ] pbx1*CLI sip show sessions [ no active SIP sessions ] Yet, pbx1*CLI core restart when convenient Waiting for inactivity to perform restart Ignoring asterisk restart request, already in progress. After doing 'core restart now' and hitting Enter really hard ;) Asterisk did restart. Some how Asterisk thinks it is not convenient. I want to find out why. -Sndr. -- | Searching for lost relatives? Win the Lottery! | 4096R/20CC6CD2 - 6D40 1A20 B9AA 87D4 84C7 FBD6 F3A9 9442 20CC 6CD2 -- _ -- 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 crashes suddenly
Quoting Daniel - Asterisk (earohua...@gmail.com): And it stops after a failed attended transfer between two of my SIP peers. May 27 09:48:32 pbx-thor-PE kernel: [334427.888524] asterisk[15384] general protection ip:482a13 sp:7f335b87c898 error:0 in asterisk[40+221000] I am using Debian 7.5 64 bits with Asterisk 11.9.0 Have you checked for core dumps? AFAIK, Debian packages create core dumps on segfaults in /tmp(?). You could use that to further pin point the issue, perhaps. $ gdb /usr/sbin/asterisk /tmp/something.core gdb bt full If it is reproducable, try recompiling Asterisk with debug symbols to get a clearer backtrace of what happened. -Sndr. -- | Do not meet girl in park, instead park meet in girl! | 4096R/20CC6CD2 - 6D40 1A20 B9AA 87D4 84C7 FBD6 F3A9 9442 20CC 6CD2 -- _ -- 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] 'restart when convenient'
Quoting Kevin Larsen (kevin.lar...@pioneerballoon.com): pbx1*CLI core restart when convenient Waiting for inactivity to perform restart Ignoring asterisk restart request, already in progress. Some how Asterisk thinks it is not convenient. I want to find out why. I haven't had it fail to restart, but I have been in the same situation and had it have a nice delay of a minute or two before it finally finds it convenient to restart. Haven't figured out what the delay is though. I am on 11.6.0. Yes. But in this case 'core show uptime' was 5 days and somewhat hours. Perhaps i should join the -dev list to find out what 'convenient' actually means for the process... -- | A good way to deal with predators is to taste terrible. | 4096R/20CC6CD2 - 6D40 1A20 B9AA 87D4 84C7 FBD6 F3A9 9442 20CC 6CD2 -- _ -- 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] 'restart when convenient'
Quoting Paddy Grice (pa...@wizaner.com): Some how Asterisk thinks it is not convenient. I want to find out why. Maybe nothing but I had a similar problem with ... when convenient - seem to remember it was a problem writing a cdr using cdr_adaptive_odbc - a database fault. Asterisk appeared idle but wouldn't close as I guess it thought something was still active. Ok. Good tip. We do use CDR in MySQL. Will check this out if it happens again! -- | And the Lord said unto John, 'Come forth and receive eternal life'. | But John came fifth and won a toaster. | 4096R/20CC6CD2 - 6D40 1A20 B9AA 87D4 84C7 FBD6 F3A9 9442 20CC 6CD2 -- _ -- 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] 'restart when convenient'
Quoting Matthew Jordan (mjor...@digium.com): The various 'stop/restart' flavours and what they mean: Thanks for this. * convenient - wait until all channels have hung up. If new channels are made, keep waiting. Once all channels have hung up, and no new channels are made, sneak in and ask all of the modules shut down and to clean up after themselves - including waiting for all CDRs to get written. This is what i have cronned atm. I found Asterisk with 5 days uptime. The behaviour i saw was issuing another 'core restart when convenient' resulting in the 'Ignoring asterisk restart request, already in progress.'-message, but issuing 'core restart now' did have effect, though, as you stated, after a short delay of about 15 seconds. I have some pointers on where to look next. It was totally unclear to me that modules 'get asked to shut down' and might not feel like doing so at particular moments. I figured it was all about active channels. Thanks. -- | If you must choose between two evils, pick the one you've never tried before. | 4096R/20CC6CD2 - 6D40 1A20 B9AA 87D4 84C7 FBD6 F3A9 9442 20CC 6CD2 -- _ -- 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 and Alcatel digital phone's
Hi Jonathan, I already looked at their product a few weeks ago, but because Alcatel wasn't on their list of compatible devices, I left it alone. Because of your email, I went looking on their site for a second time and noticed on their blog that they're experimenting with Alcatel devices. So after emailing them, there is a chance that we could use their product for our digital Alcatel phones. So fingers crossed and thanks for the info ;) Sander -Oorspronkelijk bericht- Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens Jonathan C. Bailey Verzonden: zaterdag 18 december 2010 18:19 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Asterisk and Alcatel digital phone's There is a product from Citel (the TVA) that we're currently using with Toshiba phones. I know they also support Avaya, Nortel, and Panasonic, but am not sure if they do any other brands. They more or less convert your old digital phones to SIP. They have have full compatibility information on their website... -Jon - Original Message - From: John Novack jnov...@stromberg-carlson.org To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 18, 2010 6:48:57 AM Subject: Re: [asterisk-users] Asterisk and Alcatel digital phone's Sander Naudts wrote: Asterisk and Alcatel digital phone's Hi, I'm sorry if this is already asked somewhere on the list but I couldn't find it. We have an old PBX system controlled by our Telecom provider. There are analog phones but also digital alcatel phone's connected to it. These are not ip based but legacy digital phone's. Is there a way how we can connect them to our own Asterisk PBX? The old PBX is going to be removed, so it has to be a solution: Digital alacatel phone - directly connected to Asterisk. Short answer NO! What you are calling legacy digital phones are not universal, and for many years have been integrated with the host system. This is generally true for business systems from 2 lines and six stations to large systems with hundreds of phones. Is there some hardware gateway or something we can use? The only gatewaywill be your existing switch or another of the same generation. When the switch is removed, why would the phones not be? the analog phones, if they are not special, but POTS phones that could be used anywhere on a loop start line in a business or home could be reused, but you may find that you will not want to. We looked at the Grandstream GXW4024 gateway for our analog phones but I'm not sure the digital one's can connect to that one as well. No they cannot. Better plan on replacing all the Alcatel phones with IP ones. John Novack Kind regards, Sander Naudts -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 and Alcatel digital phone's
Hi, I'm sorry if this is already asked somewhere on the list but I couldn't find it. We have an old PBX system controlled by our Telecom provider. There are analog phones but also digital alcatel phone's connected to it. These are not ip based but legacy digital phone's. Is there a way how we can connect them to our own Asterisk PBX? The old PBX is going to be removed, so it has to be a solution: Digital alacatel phone - directly connected to Asterisk. Is there some hardware gateway or something we can use? We looked at the Grandstream GXW4024 gateway for our analog phones but I'm not sure the digital one's can connect to that one as well. Kind regards, Sander Naudts -- _ -- 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] Cascading queues calls not joining unavailable queues.
Quoting Mark Michelson ([EMAIL PROTECTED]): -- Called SCCP/231 -- Called SCCP/220 -- SCCP/220-009b is busy -- SCCP/231-009a is busy I'd like asterisk to quit trying when all agents are busy, but i don't think it's possible without scripting it yourself with some AGI-script that checks 'show queues' output. It sounds as though skinny devices may not be reporting their device state correctly, and so the queue believes that the devices are available. Looking at the output of 'show queues' everything looks completely OK when i put the phone in various states of 'being available'. I think it's more an opinion on what 'unavailable' is. Or perhaps they are reporting a state that the queue does not know about. If this is the case, we may be dealing with a bug. I will test locally when I can get access to a Skinny phone and see what's going on. We're using chan_sccp.so in combination with Cisco 796x phones (With CTU ringtone! Whee! :P). Maybe it doesn't really work right because of this, but as Asterisk *tells me* it knows nobody is answering a queue, i wonder why it keeps trying ;-) Kind regards, Sander. -- | If you jog backwards, will you gain weight? | 1024D/08CEC94D - 34B3 3314 B146 E13C 70C8 9BDB D463 7E41 08CE C94D ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cascading queues calls not joining unavailable queues.
Quoting James FitzGibbon ([EMAIL PROTECTED]): Unfortunately, the patches weren't done against trunk or the head of 1.4, and the author didn't file a disclaimer with Mantis, so the bug ( http://bugs.digium.com/view.php?id=9165) was recently closed. That's just too bad, as this might be a solution to our 'problems'. :) -- | The less hair I have, the more head I get! | 1024D/08CEC94D - 34B3 3314 B146 E13C 70C8 9BDB D463 7E41 08CE C94D ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cascading queues calls not joining unavailable queues.
Quoting Mark Michelson ([EMAIL PROTECTED]): | app_queue.c: No one is answering queue '511' (7/2/0) Have you added additional queue members besides the ones you specified in queues.conf? Yes. There's a number of dynamic members that logged in to the group by means of dialing extension '*511' which will call AddQueueMember() through the dialplan. But those members were all set to DND or were otherwise engaged at the time i got the logging information i sent in my original post. I tried again with a new queue, same config, but *only* the two members specified as member = and i got this: -- Called SCCP/231 -- Called SCCP/220 -- SCCP/220-009b is busy -- SCCP/231-009a is busy -- Called SCCP/231 -- Called SCCP/220 -- SCCP/220-009d is busy -- SCCP/231-009c is busy -- Stopped music on hold on SIP/10.50.0.2-082155b8 -- Playing 'queue-youarenext' (language 'nl') -- Told SIP/10.50.0.2-082155b8 in test1 their queue position (which was 1) -- Playing 'queue-thankyou' (language 'nl') -- Started music on hold, class 'default', on SIP/10.50.0.2-082155b8 -- Called SCCP/231 -- Called SCCP/220 -- SCCP/220-009f is busy -- SCCP/231-009e is busy I'd like asterisk to quit trying when all agents are busy, but i don't think it's possible without scripting it yourself with some AGI-script that checks 'show queues' output. Any ideas? -Sndr. -- | One nice thing about egotists: They don't talk about other people. | 1024D/08CEC94D - 34B3 3314 B146 E13C 70C8 9BDB D463 7E41 08CE C94D ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cascading queues calls not joining unavailable queues.
Hi! - Trying a repost, my first message didn't seem to make the list. I have one main queue with agents that take calls to our main phonenumber. Now i want to cascade calls through to the fallback queue immediately when all the agents in the first queue are 'unreachable' in any way (be it OffHook, DND, Paused, etc...) Somehow calls still keep hanging around in the main queue even if agents are Busy or 'DND' for the specified timeout before returning to the dialplan which then calls the next queue. The extensions.conf section that places the call on the main queue and afterwards the second queue: | exten = 511,n,Queue(511,t,,,30) ; Main queue | exten = 511,n,Queue(611,t,,,30) ; Fallback queue And this is my queues.conf accordingly. I'll only show the main queue config, as the fallback queue config is EXACTLY the same, except for the queuemembers ofcourse. | [511] | servicelevel = 30 | announce = voice/connected | musiconhold = default | strategy = ringall | context = vanuit-queues | timeout = 10 | wrapuptime = 10 | announce-frequency = 10 | announce-holdtime = no | joinempty = strict | leavewhenempty = yes | member = SCCP/206 | member = SCCP/210 This selection of loglines shows Asterisk is aware that noone is answering the queue: | logger.c: -- Goto (groepen,511,1) | logger.c: -- Called SCCP/210 | logger.c: -- Called SCCP/206 | logger.c: -- SCCP/206 is busy | logger.c: -- SCCP/210 is busy | app_queue.c: No one is answering queue '511' (7/2/0) | logger.c: -- Stopped music on hold on SIP/10.10.1.1 | logger.c: -- Told SIP/10.10.1.1 in 511 their queue position (which was 1) | logger.c: -- Started music on hold, class 'default', on SIP/10.10.1.1 | logger.c: -- Called SCCP/210 | logger.c: -- Called SCCP/206 | logger.c: -- SCCP/206-0021 is busy | logger.c: -- SCCP/210-0020 is busy | app_queue.c: No one is answering queue '511' (7/2/0) | [ ... etc ... ] What am i doing wrong here? Can anyone shed some light? Thanks! Sander. -- | The short fortune teller who escaped from prison: a small medium at large. | 1024D/08CEC94D - 34B3 3314 B146 E13C 70C8 9BDB D463 7E41 08CE C94D -- | The story of my life; warm beer and cold women. | 1024D/08CEC94D - 34B3 3314 B146 E13C 70C8 9BDB D463 7E41 08CE C94D ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ping
Quoting Doug Lytle ([EMAIL PROTECTED]): Pong The list seems to act weird. I mailed to the list earlier, the message was accepted, but does not appear on the archives nor did i get a bounce or my own listmail back. Though i do see other people posting :/ -- | Only those who will risk going too far, can possibly find out how far you can go. | 1024D/08CEC94D - 34B3 3314 B146 E13C 70C8 9BDB D463 7E41 08CE C94D ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ping
Quoting Jared Smith ([EMAIL PROTECTED]): *nods* I verified more than once and even copied + pasted to make sure. Obviously my ping message went through, but my others have not. I'm working with Digium's IT department to try to track down the problem. As it may help you follow the message i 'lost' through your systems: It had Message-ID header [EMAIL PROTECTED] Your server lists.digium.com.s8a1.psmtp.com [64.18.7.10] accepted it with a TLS connection but did not reply with a queue message-id. HTH, Sander. -- | If you look like your passport picture, you probably need the trip. | 1024D/08CEC94D - 34B3 3314 B146 E13C 70C8 9BDB D463 7E41 08CE C94D ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Registry problems
Hi there I am trying to register 2 servers using iax server a and server b server a registers to server b but when i say iax2 show registry i can see it is not using port 4569 xxx.xxx.xxx.xxx:4569 123456 xxx.xxx.xxx.xxx:1024 60 Registered and now i can't register server b to server aall ports are open on the router but still timeouts what i can do is use port 1024 to register to server a but that port is changing from time to time :( i am confused ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Date based context inclusion
This should work include = day|09:00-19:59|mon|*|* include = day|09:00-19:59|fri|*|* -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Alessio Focardi Verzonden: maandag 26 september 2005 11:36 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] Date based context inclusion Hi, I know that writing in the dialplan include = day|09:00-19:59|mon-fri|*|* day will be include monday TO friday What is needed to include day monday AND friday ? include = day|09:00-19:59|mon,fri|*|* does not work, but it was just my guess Tnx for any help -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queues
Hi there i need to know if there is a wayto play a ringing sound to acallerthe enters a queue so i don't want to have music onhold and i need it to bebehind the answer option like this exten =1,1,Dial(sip/10,10) exten =1,2,Answer exten =1,3,Queue(test) thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queues
Oh thanks i looked over the r option for queues :) -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Doug Lytle Verzonden: vrijdag 23 september 2005 13:39 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Queues Sander wrote: Hi there i need to know if there is a way to play a ringing sound to a caller the enters a queue so i don't want to have music onhold and i need it to be behind the answer option like this exten =1,1,Dial(sip/10,10) exten =1,2,Answer exten =1,3,Queue(test) How about having the SIP phone a member of the test queue and have the queue ring? exten = 1,1,Answer() exten = 1,2,Queue(test|r) exten = 1,3,Hangup() exten = h,1,NoOP(Hungup) Check out this link: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Queue Doug -- Ben Franklin quote: Those who give up essential liberties for temporary safety deserve neither liberty nor safety. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco Ip phones
Hi there does any of you use ip phones from cisco on asterisk and how is the quality of this configuration ? i have to make a price of an asterisk server with 100 ip phones but i need stable phones snom is nice but still i have trouble with echo on them and budgetone is cheap and feels cheap thanks Sander ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco Ip phones
We have tested this phone with a Asterisk system and deliver the phone with pre installed SIP-firmware without License What about the license?? And do you have to buy a license and changing the phone to sip protocol looks scary :( and time consuming with 100 phones not all suppliers will do it for you, and does any of you know a supplier in the netherlands with good pricing neonova is way too expensive -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Michiel van Baak Verzonden: dinsdag 20 september 2005 20:57 Aan: asterisk-users@lists.digium.com Onderwerp: Re: [Asterisk-Users] Cisco Ip phones On 20:38, Tue 20 Sep 05, Florian Overkamp wrote: Hi Sander, Sander wrote: Hi there does any of you use ip phones from cisco on asterisk and how is the quality of this configuration ? i have to make a price of an asterisk server with 100 ip phones but i need stable phones snom is nice but still i have trouble with echo on them and budgetone is cheap and feels cheap Cisco phones work fine using SIP, good reports have also been seen with SCCP/Skinny, although my own experience on that is limited. We use SwissVoice a lot and others have reported great success with Polycom. I been using some Cisco phones for a while now. I started with converting them to SIP so they could connect to * Now with chan_sccp I reverted them all back to SCCP and they work awesome. Too bad they are so darn expensive, otherwise I wouldn't use anything else. Just my experience :) -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco Ip phones
I have a snom 360 installed but the woman that is operating it complains about it all the time i looked at it and sometimes when sh transfers a phonecall it will just hang and stays in the phone the snom does not have connection to the line you can only see the line is still there in the display it tells you connected i think it's something like she don't push the buttons in good enough. But they complain about many things mostly they have to look inside there company phones are ringing but nobody answers them :) -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Michiel van Baak Verzonden: dinsdag 20 september 2005 22:01 Aan: asterisk-users@lists.digium.com Onderwerp: Re: [Asterisk-Users] Cisco Ip phones On 21:30, Tue 20 Sep 05, Anders Svensson wrote: Have you tested Aastra. Works great with * and reasoable pricing Nope, haven't seen any phone of them in real life yet. Right now we deploy snom's for the price/quality rate they deliver. I find them very stable and nice phones. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to create IVR menu and transfer to anothersip extensions.
Thisisverybasicprogrammingandisexplainedinatutorials is you have a sip phone you will have a transfer or flash button so you can transfer any call to another a ivr menu is very simple to exten = s,1,Answer exten = s,2,Background(audiofile..) exten = 1,1,Dial(sip/100) exten =2,1,Dial(sip/101) press 1 to dial extension 100 press 2 to dial extension 101 it's as easy as thatand if you want an example you have to provide more details http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/book1.htmlread this link and you will know all the basics Good luck Sander Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens PJ SantosVerzonden: woensdag 14 september 2005 18:16Aan: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial DiscussionOnderwerp: Re: [Asterisk-Users] How to create IVR menu and transfer to anothersip extensions. I need create one configuration to provide one Interactive Voice Response. I read any docs about this. So, if you have one sample, please post. Thanks. Paulo Santos. Brasil-RJMoises Silva [EMAIL PROTECTED] escreveu: mmm actually i think that is a functionality most VoIP phones provide, you dont need do anything, just press transfer in your VoIP phone and the dial the extension you want to transfer to. On 9/13/05, PJ Santos [EMAIL PROTECTED] wrote: Hi All, I need help to create one IVR Menu, when a say "Welcome to PBX Corp..." , press 1 to Sales, press 2 to Help Desk or wait to operator. What function should I use for call transfer exten SIP to exten SIP. eg I call to extension 190 and after answer, I do one transfer to another exten SIP. Regards. Paulo Santos Yahoo! Messenger com voz: PROMOÇÃO VOCÊ PODE LEVAR UMA VIAGEM NA CONVERSA. Participe! ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org " ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger com voz: PROMOÇÃO VOCÊ PODE LEVAR UMA VIAGEM NA CONVERSA. Participe! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] timeout with queue
In queues.conf ; How long do we let the phone ring before we consider this a timeout... ; timeout = 15 But this is just the function how long the phones will ring you should not set this option to long or your phone will stop ringing if a timeout is set in your phone But when the line hangs up after timeout you have set an option at the queue like this below it will stay in queue for 15 seconds then hangs up exten = 121,2,Queue(121|tT|||15) Exten = 121,3,Hangup -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Wolfgang Lumpp Verzonden: woensdag 14 september 2005 16:25 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] timeout with queue Hi, I've setup a queue with 3 sip members. I've tried with random and roundrobin and different timeout settings in musiconhold.conf Always after the second Nobody picked up in 15000ms I get Exiting on time-out cycle Stopped music on hold on CAPI/contr1/s-0 Where can I increase this timeout? asterisk 1.0.9 on linux 2.6.11 SuSE 9.3 Thanks a lot Regards Wolfgang ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: (no subject)
This is not a siemens pbx problem you set the
pridialplan = to national and that adds a number to the outgoing call or
something just use
Pridialplan = local
prilocaldialplan = local
and it should work
I tried to open the file kds again and now it showed me your configuration
:) don't know why it did not show me before
Sander
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Pablo Allietti
Verzonden: woensdag 14 september 2005 17:31
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: [Asterisk-Users] Re: (no subject)
On Wed, Sep 14, 2005 at 10:22:10AM -0400, Matt Ryanczak wrote:
ok. didnt work :( i thinks is a pbx problem. because E1 is incomming in the
pbx. and all incomming calls go to 100. thats the problem i will try to
solve this.
It could potentially be both. I would look at your extensions.conf
first though. What does the extension entry for that context look like.
For instance I have an entry in my extensions.conf for dialing outside
lines (outside being from asterisk to my PBX and then onto the outside
world from there). The entry looks like this:
[to-analog]
exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN}) exten = _9XXX.,2,Congestion
exten = _9XXX.,103,Hangup
To dial a PBX extension the entry would look almost the same:
[to-pbx-extension]
exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN:1}) exten = _9XXX.,2,Congestion
exten = _9XXX.,103,Hangup
Hope this helps,
-Matt
On Wed, 2005-09-14 at 11:46 -0300, Pablo Allietti wrote:
hi all, i have a box with a te110p and a pbx siemens... connect both
with a e1.
with a xten soft i can call extensions numbers in my office example
100
102 etc. but when i truy to go outside with the 9 before the call
rings in the first extensions (100). this is a asterisk problem? or
a pbx problem?
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RE: [Asterisk-Users] Re: (no subject)
Ok it was a problem with my provider it could not see the right numbers
comming in :)
You can start maintenence in the manager e tool from siemens and start a
trace or start call monitoring on extension 100 then you can see the number
the asterisk has given to the pbx to dial.
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Pablo Allietti
Verzonden: woensdag 14 september 2005 21:17
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: [Asterisk-Users] Re: (no subject)
On Wed, Sep 14, 2005 at 07:52:26PM +0200, Sander wrote:
This is not a siemens pbx problem you set the pridialplan = to
national and that adds a number to the outgoing call or something just
use
Pridialplan = local
prilocaldialplan = local
and it should work
no uuuaaa the same problem.. ring in the extension 100.
I tried to open the file kds again and now it showed me your
configuration
:) don't know why it did not show me before
Sander
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Pablo Allietti
Verzonden: woensdag 14 september 2005 17:31
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: [Asterisk-Users] Re: (no subject)
On Wed, Sep 14, 2005 at 10:22:10AM -0400, Matt Ryanczak wrote:
ok. didnt work :( i thinks is a pbx problem. because E1 is incomming
in the pbx. and all incomming calls go to 100. thats the problem i
will try to solve this.
It could potentially be both. I would look at your extensions.conf
first though. What does the extension entry for that context look like.
For instance I have an entry in my extensions.conf for dialing
outside lines (outside being from asterisk to my PBX and then onto
the outside world from there). The entry looks like this:
[to-analog]
exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN}) exten = _9XXX.,2,Congestion
exten = _9XXX.,103,Hangup
To dial a PBX extension the entry would look almost the same:
[to-pbx-extension]
exten = _9XXX.,1,Dial(ZAP/G1/${EXTEN:1}) exten =
_9XXX.,2,Congestion exten = _9XXX.,103,Hangup
Hope this helps,
-Matt
On Wed, 2005-09-14 at 11:46 -0300, Pablo Allietti wrote:
hi all, i have a box with a te110p and a pbx siemens... connect
both with a e1.
with a xten soft i can call extensions numbers in my office
example 100
102 etc. but when i truy to go outside with the 9 before the call
rings in the first extensions (100). this is a asterisk problem?
or a pbx problem?
___
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---end quoted text---
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Pablo Allietti
LACNIC
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---end quoted text---
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RE: [Asterisk-Users] Starting From Scratch
Try this just dial 0 to dial out and then dial a internal number at you
company
What are the between 916 and 2128 ?
exten = _0.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN},70,Tt)
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Thczv F. Thczv
Verzonden: donderdag 15 september 2005 5:55
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: [Asterisk-Users] Starting From Scratch
Hello all:
For fun, I am learning about Asterisk, and trying to get Asterisk working at
my house. I installed [EMAIL PROTECTED] It seems to be functioning fine. I
installed a couple of softphones, and have them registered with Asterisk. I
actually work for a CLEC, and I have registered my Asterisk box with SER
(which I don't begin to understand
yet) at the office. In order to try to understand how all this works, I
have stripped my extensions.conf down to almost nothing. I am building it
up piece by piece. This is the entirety of my extensions.conf file:
[globals]
OUTBOUNDTRUNK=SIP/mysipprovider.com
[from-internal]
exten = 105,1,Answer()
exten = 105,2,Playback(abandon-all-hope) exten = 105,3,Hangup() exten =
106,1,Dial(${OUTBOUNDTRUNK}/916xxx6000)
exten = 107,1,Dial(${OUTBOUNDTRUNK}/916xxx2128)
This is all just testing. When I dial 105 from either of my softphones, it
plays the recording fine. My thought for the 106 and
107 extensions was to sort of hard code it so that if I dialed either of
those extensions the call would automatically get routed over my outbound
sip trunk to the appropriate offsite PSTN dialable number.
Once I know that the calls will go through, I will create a proper dial
plan.
The 6000 number is my home PSTN phone. The 2128 number is my office desk
phone (a SIP phone). Here is where I get stumped: When I dial
107 from my SIP softphone, the call goes out fine, and rings my SIP office
phone just fine. In other words, my asterisk box passes the call to my
company's server, and it goes through. The guys at work tell me that
outside PSTN calls (like to my home PSTN phone) should work exactly the same
way (no special dialing needed, just the standard 10 digit telephone
number). But for some reason when I dial
106 from my softphone it doesn't work. I get a recording that tells me the
person you are trying to reach is unavailable.
As near as I can tell (as someone unexperienced with this) from looking at
the text that gets spit out when I run SIP debug, both calls go through
the same. The debug info from when I dialed 106 is included below.
I know I am a dummy about this stuff. But I am trying to learn how it
works. Have mercy on me you experts. Can any of you see what might be
wrong?
My guesses include two possibilities:
1. For some reason my PSTN onramp (my own company) isn't really passing the
call to the PSTN as it should.
2. Even for this really basic hardcoding experiment, my extensions.conf
file is too short. For example, the connection takes longer than asterisk
expects, and so I need to tell it to keep waiting before playing the
recording. Or something like that.
Any ideas?
Thanks,
Dave
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK032524f5
From: 102 sip:[EMAIL PROTECTED];tag=as0dbb6283
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 15 Sep 2005 03:41:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 1541 1541 IN IP4 192.168.1.200
s=session
c=IN IP4 192.168.1.200
t=0 0
m=audio 17464 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to 66.81.0.87:5060
-- Called smf-reg.sip.o1.com/916xxx6000 asterisk1*CLI
Sip read:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bK032524f5;received=24.23.48.16
From: 102 sip:[EMAIL PROTECTED];tag=as0dbb6283
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Server: Sip EXpress router (0.8.14 (i386/linux))
Content-Length: 0
Warning: 392 66.81.0.87:5060 Noisy feedback tells: pid=22068
req_src_ip=24.23.48.16 req_src_port=5060
in_uri=sip:[EMAIL PROTECTED]
out_uri=sip:[EMAIL PROTECTED] via_cnt==1
9 headers, 0 lines
asterisk1*CLI
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.200:5060;received=24.23.48.16;branch=z9hG4bK032524f5
Record-Route: sip:[EMAIL PROTECTED];r2=on;ftag=as0dbb6283;lr=on
Record-Route: sip:[EMAIL PROTECTED];r2=on;ftag=as0dbb6283;lr=on
From: 102 sip:[EMAIL PROTECTED];tag=as0dbb6283
To: sip:[EMAIL PROTECTED];tag=as5f7d7868
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk SIPv2 (http://www.asterisk.org
CVS-HEAD-03/02/05-12:13:56 )
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Type: application/sdp
Content-Length: 210
v=0
o=root 9338
RE: [Asterisk-Users] Integration between Asterisk and Siemens HiCom150e over ISDN
Just setup the stls4 card to work in NT mode and connect the siemens pbx to the asterisk with a crossover cable. Then you will be able to make calls to from the hicom to the asterisk machine you do not need to have an nt box to make the connection. With the nt box you can power an ISDN phone if the phone needs power. -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Simon D Verzonden: dinsdag 13 september 2005 15:14 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] Integration between Asterisk and Siemens HiCom150e over ISDN Hi, I am looking to integrate Asterisk with a Siemens HiCom 150e via BRI and wondered if anyone is able to offer any advice. In simplistic terms, my goal is to pass calls from the HiCom to the Asterisk box. e.g: HiCom user dials access code and can call Asterisk extension or establish SIP call over Internet. Likewise, I'd like Asterisk to be able to present a call to the Hicom, either Asterisk extension calling HiCom extension, or an incoming Sipgate call presented to the HiCom, for example. My hardware: - Asterisk: I have an Asterisk box configured with 1x Sitecom DC105 PCI ISDN Card (HFC chipset, TE/NT capable). [and 2x X100P Analogue FXOs, but that's not relevant here] My understanding is that I should configure the ISDN card in NT mode and power the bus with an NT1, or will a crossover cable in to the HiCom suffice? Reference to NT1's and line power here: http://isdn.jolly.de/download/v3.0/PBX4Linux-3.0.pdf - Siemens HiCom 150e: I currently have a HiCom 150e switch with digital and analogue stations and an analogue trunk card (TLA4). I also have an STLS4 card. Initially, I thought this would be the answer to my prayers but now am not so sure... According to http://www.webco.com/siemens/interfaces.html, my STLS4 is defined as the following: Connects up to 4 ISDN S0 terminals (8 channels) for data equipment and video conferencing applications to the OfficePoint or OfficeCom. Connected devices must provide their own power. Some special wiring for the ISDN S0 device connection is required. Also requires connection of ISDN BRI trunks (TMQ4) for network access. Documentation on ISDN BRI: http://web2.tac.siemenscom.com/pub/150e/Config/Note020.pdf Documentation on ISDN S0 Device Install: http://web2.tac.siemenscom.com/pub/150e/Config/Note009.pdf I can't figure out if the SLTS4 is the correct card for my requirements, or do I need a TMQ4 *as well* or a TMQ4 *instead*?.. Trawling through previous posts, I've found reference to the STLS4 here: http://lists.digium.com/pipermail/asterisk-users/2005-January/081449.html Is anyone able to help? Many thanks in advance, Simon England, UK ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoipBuster again
Try this ip for register something looks wrong with iax.voipbuster.com
I changed it a while ago because i had some dns problems in with my provider
and this ip came up when i pinged now you can't ping to the adress and it's
another ip
register = username:[EMAIL PROTECTED]
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Rudolf Ladyzhenskii
Verzonden: zondag 11 september 2005 0:25
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [Asterisk-Users] VoipBuster again
Here is what I get when reloading IAX2:
Not every time, though
== Parsing '/etc/asterisk/iax.conf': Found Sep 11 08:48:29 WARNING[3240]:
chan_iax2.c:5402 iax2_register: Host 'iax.voipbuster.com' not found at line
164
Strange, because name resolves to IP address.
Ok, I reload IAX2 again and no more warning.
Then it tries to register and fails:
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: LAGRQ
Timestamp: 10018ms SCall: 1 DCall: 0 [213.61.187.146:4569]
Tx-Frame Retry[001] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: LAGRQ
Timestamp: 10018ms SCall: 1 DCall: 0 [213.61.187.146:4569]
Tx-Frame Retry[002] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ
Timestamp: 00017ms SCall: 1 DCall: 0 [213.61.187.146:4569]
USERNAME: USERNAME
REFRESH : 60
And so it goes.
Call then fails too...
I am suspecting two things:
1. I am starting to wonder if registering a user in Australia using
VoipBuster application does not create an IAX account
Can someone who has an IAX account try creating one for me? Bogus name and
password. my e-mail is [EMAIL PROTECTED]
2. Firewall ports are not open. I am sure all the right ports are forwarded
to my * box (5060, 4569, 1-2).
I will set up ethereal on my firewallbox to see what comes out to the www
and what comes back.
Thanks,
Rudolf
- Original Message -
From: Sander [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Saturday, September 10, 2005 11:32 PM
Subject: RE: [Asterisk-Users] VoipBuster again
Iax.conf
register = username:[EMAIL PROTECTED]
Extensions.conf
exten =
_0.,2,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN:\1\},\
60,r)
Good luck :) Sander
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Rudolf
Ladyzhenskii
Verzonden: zaterdag 10 september 2005 13:57
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: [Asterisk-Users] VoipBuster again
Hi, all
I am still battling to connect * and voipbuster.
What protocol does it use? Ethereal capture shows UDP traffic, but no SIP
or
IAX traffic when using their client.
VoipBuster client connects to connectionserver.voipbuster.com on port
2
for authentication. Call itself is placed on different server.
I have tried to connect using SIP and IAX and it seems that no
authentication is happening. (i was trying to use sip.voipbuster.com and
iax.voipbuster.com). Does anyone currently use asterisk with voipbuster?
If
so, can you you help me to set it up? I am really lost.
My setup is :
sip.conf
[voipbuster]
type=peer
insecure=very
host=sip.voipbuster.com
username=NAME
secret=SECRET
fromdomain=sip.voipbuster.com
realm=voipbuster.com
iax.conf:
[voipbuster]
type=peer
host=iax.voipbuster.com
username=NAME
secret=NAME
notransfer=yes
qualify=no
extensions.conf: (I use 0 to dial out to IAX and 8 to dial out SIP) exten
=
_0.,1,SetCallerID(CID Name CIDNUMBER) exten =
_0.,2,Dial,IAX2/voipbuster/00613${EXTEN:1}
exten = _8.,1,SetCallerID(CID Name CIDNUMBER) exten =
_8.,2,Dial,SIP/voipbuster/00613${EXTEN:1}
Thanks,
Rudolf
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RE: [Asterisk-Users] Configuring SIPURA 2002 to work wih Asterisk
you can try to post your sip.confso someone can help the sipura spa 2002 works perfectly with asterisk Sander Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Paul ConnVerzonden: zaterdag 10 september 2005 23:15Aan: asterisk-users@lists.digium.comOnderwerp: [Asterisk-Users] Configuring SIPURA 2002 to work wih Asterisk Im setting up Asterisk for the first time. I purchased a SIPURA 2002 ATA to connect with the Asterisk server. In the /var/log/asterisk/messages log I keep getting an error indicating wrong password. Below is the error I am receiving. Note that the IP address and username has been modified for security. Sep 10 15:56:22 NOTICE[24099] chan_sip.c: Registration from 'John Doe sip:[EMAIL PROTECTED] ' failed for '192.168.1.5' - Wrong password In the sip.conf file under the extensions I have the secret set the same way as the password in the SIPURA 2002 GUI under the LINE 1 parameters. Anyone successfully configured the SIPURA 2002 to work with Asterisk OR does anyone know of any help documents (other than the SIPURA PDF) that explains the configuration of the 2002 for use with asterisk? Thanks! Paul ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoipBuster again
Iax.conf
register = username:[EMAIL PROTECTED]
Extensions.conf
exten = _0.,2,Dial(IAX2/username:[EMAIL PROTECTED]/${EXTEN:\1\},\
60,r)
Good luck :) Sander
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Rudolf Ladyzhenskii
Verzonden: zaterdag 10 september 2005 13:57
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: [Asterisk-Users] VoipBuster again
Hi, all
I am still battling to connect * and voipbuster.
What protocol does it use? Ethereal capture shows UDP traffic, but no SIP or
IAX traffic when using their client.
VoipBuster client connects to connectionserver.voipbuster.com on port 2
for authentication. Call itself is placed on different server.
I have tried to connect using SIP and IAX and it seems that no
authentication is happening. (i was trying to use sip.voipbuster.com and
iax.voipbuster.com). Does anyone currently use asterisk with voipbuster? If
so, can you you help me to set it up? I am really lost.
My setup is :
sip.conf
[voipbuster]
type=peer
insecure=very
host=sip.voipbuster.com
username=NAME
secret=SECRET
fromdomain=sip.voipbuster.com
realm=voipbuster.com
iax.conf:
[voipbuster]
type=peer
host=iax.voipbuster.com
username=NAME
secret=NAME
notransfer=yes
qualify=no
extensions.conf: (I use 0 to dial out to IAX and 8 to dial out SIP) exten =
_0.,1,SetCallerID(CID Name CIDNUMBER) exten =
_0.,2,Dial,IAX2/voipbuster/00613${EXTEN:1}
exten = _8.,1,SetCallerID(CID Name CIDNUMBER) exten =
_8.,2,Dial,SIP/voipbuster/00613${EXTEN:1}
Thanks,
Rudolf
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RE: [Asterisk-Users] Huge Echo
Your config files?? Zaptel zapata ? Maybe you have a setting wrong i had the same problem but then with pri lines now it's gone. You can hear yourself as loud as the other person that is calling you? And what sipphone do you use -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Marek Zachara Verzonden: vrijdag 9 september 2005 13:27 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Huge Echo On Friday 09 of September 2005 13:14, Andreas Sikkema wrote: [EMAIL PROTECTED] wrote: In the following setup: call coming from a pstn line - into FXO card - asterisk - SIP phone i get an incredible loud echo in the SIP phone (about 0,5-1s) (everything i speak into SIP phone microphone i hear in its speaker). The person calling from PSTN is not getting any echo. Make sure you're not playing the recorded sound from your microphone back to your loudspeakers. How could I have done that? I'm not recording any sound (at least nothing i'm aware of). The echo doesn't happen when the call is incoming from SIP provider (instead of PSTN) - so i assume the problem is related to the analog line. The SIP phone is stand-alone AT-320 Marek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] pri gateway
Not all providers use crc4 you can try to remove the entry -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens altus Verzonden: vrijdag 9 september 2005 7:24 Aan: Baris Simsek CC: asterisk Onderwerp: Re: [Asterisk-Users] pri gateway These are my configs for a sangoma 4 port connected to E1's in the UK loadzone = us loadzone=uk defaultzone=uk span=1,1,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 span=3,1,0,ccs,hdb3,crc4 span=4,1,0,ccs,hdb3,crc4 # card 0 - span 1 bchan=1-15,17-31 dchan=16 # card 0 - span 2 bchan=32-46,48-62 dchan=47 # card 0 - span 3 bchan=63-77,79-93 dchan=78 # card 0 - span 4 bchan=94-108,110-124 dchan=109 and zapata.conf [channels] context=default switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown priindication = outofband usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=yes callwaitingcallerid=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes callgroup=1 pickupgroup=1 ; card 0 - span 1 switchtype = euroisdn signalling = pri_cpe group = 1 context = incoming channel = 1-15,17-31 ; card 0 - span 2 switchtype = euroisdn signalling = pri_cpe group = 1 context = incoming channel = 32-46,48-62 ; card 0 - span 3 switchtype = euroisdn signalling = pri_cpe group = 1 context = incoming channel = 63-77,79-93 ; card 0 - span 4 switchtype = euroisdn signalling = pri_cpe group = 1 context = incoming channel = 94-108,110-124 Maybe its your telco?? On Thu, 2005-09-08 at 15:23 +0300, Baris Simsek wrote: hi, my asterisk version is 1.0.9 /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 it is comfortable with Turkish Telecom. i tried before and it works. /etc/asterisk/zapata.conf [channels] switchtype=euroisdn signalling=pri_cpe context=incoming group=1 channel=1-15,17-31 Leds are lighting at start. When i run /etc/init.d/zaptel they go out. And i can see the modules are installed. and i see that, layer 1 is going up after zaptel. So i am sure there is no problem with drivers. I think it is connected to asterisk. any idea? thanks... altus wrote: what about a copy of your zapata.conf and zaptel.conf,what color is the leds On Thu, 2005-09-08 at 12:42 +0300, Baris Simsek wrote: hello, i installed an asterisk as a pri gateway. Everything is okay. /etc/init.d/zaptel starts successfull with wct4xxp module. /etc/init.d/asterisk starts also successfully. I tested my pri cable and it works. But still my span isn't up. I don't see any error. Do you have any idea? What else i should check? Thanks. My card is 4 span Wildcard TE410P http://www.digium.com/index.php?menu=product_detailcategory=hardwa reproduct=TE410P # lsmod wct4xxp 106688 62 zaptel226820 129 wct4xxp # asterisk -r gw*CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, In Alarm, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] siemens pbx what i ask techinician?
It's not that easy then everytime you want to change someting for testing you have to ask them to change something i can give you the software for programming siemens pbx if you want -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Pablo Allietti Verzonden: vrijdag 9 september 2005 16:09 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] siemens pbx what i ask techinician? im really newbie, and i have a siemens digital pbx work in my work. i have 4 outside lines and the pbx has a E1/PRI card. what i need to ask my siemens provider(techinicians) to do in the pbx? i only have in my pbx the 9 to get a line to go outside is very simple. but i dont know what i need to ask them to programming. please help me. -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: siemens pbx what i ask techinician?
Do you want to connect the asterisk with pri or with internal isdn? And what model pbx do you have? then i can tell you how to configure? Maybe some screenshots with it -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Pablo Allietti Verzonden: vrijdag 9 september 2005 19:35 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] Re: siemens pbx what i ask techinician? On Fri, Sep 09, 2005 at 05:39:22PM +0200, Sander wrote: thanks Sander but i have the soft, and i can enter to the pbx conf and modify all settings, but i dont know how settings i need to change. It's not that easy then everytime you want to change someting for testing you have to ask them to change something i can give you the software for programming siemens pbx if you want -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Pablo Allietti Verzonden: vrijdag 9 september 2005 16:09 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] siemens pbx what i ask techinician? im really newbie, and i have a siemens digital pbx work in my work. i have 4 outside lines and the pbx has a E1/PRI card. what i need to ask my siemens provider(techinicians) to do in the pbx? i only have in my pbx the 9 to get a line to go outside is very simple. but i dont know what i need to ask them to programming. please help me. -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: siemens pbx what i ask techinician?
Ok i'm looking will try to make a small manual for you, please make sure you have set the jumers of the pri card in asterik at the right position? Voor TE mode Here is my config of the e1 card the only thing that does not work on my pbx is to do a reboot with de cable plugged in the asterisk pbx don't know why but it just hangs. so boot after power down Zaptel: loadzone=us defaultzone=us # E1 definition: crc4 check achter hdb3 ,crc4 span=1,1,0,ccs,hdb3 #E1: bchan=1-15,17-31 dchan=16 zapata [channels] signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=no callerid=asreceived group=1 immediate = no context= incomming ; Points to the default context of your extensions.conf channel = 1-15,17-31 And on the siemens pbx turn of crc4 check on the e1 card configuration maybe you can give me your mail adres so i can make screenshots of the manager e configuration tool i can't mail pictures to the user list -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Pablo Allietti Verzonden: vrijdag 9 september 2005 20:29 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] Re: siemens pbx what i ask techinician? On Fri, Sep 09, 2005 at 07:20:58PM +0200, Sander wrote: uuauuu that will great! i cant undertand too much about internal connection because. i have a PC with a te110p and my siemens is a hipath 3500 with a s2m/pri card is a E1 card. but i dont know how to connect between them. i have always the red alarm in the te110p. my conf files are both of this files i copy and paste from internet. /etc/zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 loadzone= us defaultzone = us and the /etc/asterisk/zapata.conf [channels] context=zap-in ;switchtype=qsig pridialplan=national signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no callprogress=no callerid=asreceived group=1 signalling=pri_net channel = 1-15,17-31 please help me!!! thanks a lot for your time Do you want to connect the asterisk with pri or with internal isdn? And what model pbx do you have? then i can tell you how to configure? Maybe some screenshots with it -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Pablo Allietti Verzonden: vrijdag 9 september 2005 19:35 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] Re: siemens pbx what i ask techinician? On Fri, Sep 09, 2005 at 05:39:22PM +0200, Sander wrote: thanks Sander but i have the soft, and i can enter to the pbx conf and modify all settings, but i dont know how settings i need to change. It's not that easy then everytime you want to change someting for testing you have to ask them to change something i can give you the software for programming siemens pbx if you want -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Pablo Allietti Verzonden: vrijdag 9 september 2005 16:09 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] siemens pbx what i ask techinician? im really newbie, and i have a siemens digital pbx work in my work. i have 4 outside lines and the pbx has a E1/PRI card. what i need to ask my siemens provider(techinicians) to do in the pbx? i only have in my pbx the 9 to get a line to go outside is very simple. but i dont know what i need to ask them to programming. please help me. -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http
RE: [Asterisk-Users] Re: siemens pbx what i ask techinician?
Oh maybe you can send me your config file from the pbx and and then i can make changes for you and check the settings you can then see what i changed? -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Pablo Allietti Verzonden: vrijdag 9 september 2005 20:29 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] Re: siemens pbx what i ask techinician? On Fri, Sep 09, 2005 at 07:20:58PM +0200, Sander wrote: uuauuu that will great! i cant undertand too much about internal connection because. i have a PC with a te110p and my siemens is a hipath 3500 with a s2m/pri card is a E1 card. but i dont know how to connect between them. i have always the red alarm in the te110p. my conf files are both of this files i copy and paste from internet. /etc/zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 loadzone= us defaultzone = us and the /etc/asterisk/zapata.conf [channels] context=zap-in ;switchtype=qsig pridialplan=national signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no callprogress=no callerid=asreceived group=1 signalling=pri_net channel = 1-15,17-31 please help me!!! thanks a lot for your time Do you want to connect the asterisk with pri or with internal isdn? And what model pbx do you have? then i can tell you how to configure? Maybe some screenshots with it -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Pablo Allietti Verzonden: vrijdag 9 september 2005 19:35 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] Re: siemens pbx what i ask techinician? On Fri, Sep 09, 2005 at 05:39:22PM +0200, Sander wrote: thanks Sander but i have the soft, and i can enter to the pbx conf and modify all settings, but i dont know how settings i need to change. It's not that easy then everytime you want to change someting for testing you have to ask them to change something i can give you the software for programming siemens pbx if you want -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Pablo Allietti Verzonden: vrijdag 9 september 2005 16:09 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] siemens pbx what i ask techinician? im really newbie, and i have a siemens digital pbx work in my work. i have 4 outside lines and the pbx has a E1/PRI card. what i need to ask my siemens provider(techinicians) to do in the pbx? i only have in my pbx the 9 to get a line to go outside is very simple. but i dont know what i need to ask them to programming. please help me. -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http
RE: [Asterisk-Users] Siupra-2002 with astersik
What is your problem with asterisk ans sipura ? Config files ?? Settings Give some more info on the problems -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Joseph Verzonden: donderdag 8 september 2005 23:18 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] Siupra-2002 with astersik Is anybody using Sipura 2002 unit with asterisk? I have a problem with outgoing calls. -- #Joseph ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] atxfer featuremap
Hi there i just can't find an answer on the featuremap config i want all phones to use the same method for transfering a call on all phones but i just can't get the atxfer or other functions to work on my grandsteam and sipura spa 2000 it's confusing for users with different phones to transfer a call i know you can use the transfer button but i wan't to use a code *1 not possible??? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura Devices and Asterisk?
hi there I'm using the sipura spa 2000 they work great and good audio quality the only thing i am having trouble with is the the second port on the sipura takes a while before ringing,sometimes after 3 rings i have 13 sipura spa 2000 on a single asterisk server ,better isa channel bank i think. Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Benjamin AmezcuaVerzonden: dinsdag 6 september 2005 22:22Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'Onderwerp: RE: [Asterisk-Users] Sipura Devices and Asterisk? Don´t look for anymore!! These are the best devices (quality/price) in the market. Ben -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowanSent: martes, 06 de septiembre de 2005 22:09To: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] Sipura Devices and Asterisk? I'm currently using the Linksys PAP2, and since there's a shortage I'm looking for different devices. I'm mainly looking at the Sipura SPA sets since they are the base of the pap2. Anyone else have experience using them, and which one? Thanks Sherwood McGowan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queue AgentCallBackLogin
Hi there you let the calls go to local then it will always go to voicemail,
exten = 1234,1,Dial(local/1234) use Dial(sip/1234)
And how do you call the queue i don't see a queue extension
exten = 12345567,1,Answer
Exten = 12345567,2,queue(test1)
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens João Paulo Antunes
Verzonden: dinsdag 6 september 2005 22:49
Aan: asterisk-users@lists.digium.com
Onderwerp: [Asterisk-Users] Queue AgentCallBackLogin
Hi All,
I'm having trouble setting up a queue: I'm using AgentCallBackLogin to login
in the queue, but:
1 - When an agent answer the call and another call arrive his phone rings
again.
2 - When no there are no one answer the queue the system goes to voicemail
of agent 1234
I'm using asterisk-1.2.0-beta1.
My configuration is below,
Any ideas?
Many thanks,
Joao Antunes
;extensions.conf
[demo]
exten = 1005,1, Answer
exten = 1005,2, AgentCallBackLogin(${CALLERIDNUM}|[EMAIL PROTECTED])
exten = 1005,3, AddQueueMember(test1|local/[EMAIL PROTECTED])
exten = 1005,n, Hangup
exten = 1006,1,Answer
exten = 1006,2, AgentCallBackLogin(${CALLERIDNUM}|'##')
exten = 1006,3,RemoveQueueMember(sporski|Local/[EMAIL PROTECTED])
exten = 1006,4, Hangup
exten = 1235,1,Dial(SIP/1235,30,,http://www.teste.pt)
exten = 1235,2,VoiceMail(u1235)
exten = 1235,3,Hangup
exten = 1234,1,Dial(SIP/1234,20)
exten = 1234,2,VoiceMail(u1234)
exten = 1234,3,Hangup
[agentes]
exten = 1234,1,Dial(local/1234)
exten = 1235,1,Dial(local/1235)
exten = 1236,1,Dial(local/1236)
;queue.conf
[test1]
music = random
musiconhold=default
strategy = ringall
announce-holdtime = once
joinempty = yes
eventwhencalled = yes
eventmemberstatusoff = no
;agents.conf
[agents]
agent = 1234,,Peter Mary
agent = 1235,,Ronald Chunk
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[Asterisk-Users] call queues problem
Hi there i just setup a asterisk box with autoattendant and call queues, but it seems that when one of the agents is busy all the new calls will stay on hold until The agent hangs up then all phone will ring [aftersales] musiconhold = default timeout = 15 retry = 5 maxlen = 0 member = sip/131 member = sip/132 member = sip/133 member = sip/134 this is the queue hope someone can help me here ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup Issues on TDM40B FXO Australia
I had a similar issue both with the X100P clones and TDM400. Both were fixed by enabling AU zone and the busydetect functions. Don't forget a full asterisk reload needs to take place after changing Zap conf files, not just a soft-reload. Best way is to reboot the computer. Mike I have a similar issue. I have 2 pstn lines and a phone plugged into my tdm400. If I make a call to the outside using the phone, and the pstn number is engaged, and I hang up, the line is not freed. I have been restarting asterisk to get my external line back. This does not happen if I make the same call from my pc (using sj phone). Malcolm [EMAIL PROTECTED] wrote: Afternoon all, After doing some test on my asterisk box I can successfully receive calls to my Asterisk PBX to a SIP phone from The Telstra PSTN network Dial out from a sip phone is also not an issue, all calls connect and terminate normally. If I call the Asterisk PBX say from PSTN in Zap1-1 and out through ZAP2-2 back to the PSTN (after entering the correct pin off course) the card does not appear to detect the hang-up, I then have to issues a soft hang-up to close the call, I presume this indicates the card is configured to receive the correct hangup signal I have tried enabling callprogress, busydetect and a few settings on the busycount but to no success I've also tried LS and KS signalling Does anyone else have any suggestions to get this to work with Australia's Telstra? Regards Haydn This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the originator of the message. This footer also confirms that this email message has been scanned for the presence of computer viruses. Any views expressed in this message are those of the individual sender, except where the sender specifies and with authority, states them to be the views of LMC. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.15 - Release Date: 22/05/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music On Hold problem: Read 392 bytes of audiowhile expecting 1600
Hi there I am not an expert but you cant use variable bitrate mp3s on asterisk maybe thats the problem Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] Namens Michael Stahl Verzonden: dinsdag 17 mei 2005 23:35 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] Music On Hold problem: Read 392 bytes of audiowhile expecting 1600 My new asterisk install seems to be running fine - including playing all prompts etc without error. However, when placing someone on hold they here choppy music (first second or so) then quiet. I see the errors below. What is causing this? (Note that I am running AsteriskWin32). Thanks, Mike May 17 17:26:02 VERBOSE[3708]: -- Executing Answer(SIP/2434-263c, ) in new stack May 17 17:26:02 VERBOSE[3708]: -- Executing WaitMusicOnHold(SIP/2434-263c, 30) in new stack May 17 17:26:02 VERBOSE[3708]: -- Started music on hold, class 'default', on SIP/2434-263c May 17 17:26:02 DEBUG[3708]: Stopping retransmission on '603eb611c40e9620' of Response 2: Found May 17 17:26:02 DEBUG[3708]: Ooh, format changed from unknown to ulaw May 17 17:26:03 DEBUG[3708]: Read 392 bytes of audio while expecting 1600 May 17 17:26:04 DEBUG[3708]: Read 392 bytes of audio while expecting 1600 May 17 17:26:05 DEBUG[3708]: Read 392 bytes of audio while expecting 1600 May 17 17:26:06 DEBUG[3708]: Read 392 bytes of audio while expecting 1600 May 17 17:26:07 DEBUG[3708]: Read 394 bytes of audio while expecting 1600 May 17 17:26:08 DEBUG[3708]: Read 392 bytes of audio while expecting 1600 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SNOM190 DTMF problem
You have to set dtmfmode to rfc2833 in your sip.conf That should work I have a SNOM 360 and I have no problems at all dtmfmode=rfc2833 -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Damian Funnell Verzonden: vrijdag 13 mei 2005 0:59 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] SNOM190 DTMF problem Hi all, We've got a problem where a bunch of SNOM 190 phones that we have just installed are giving us problems with DTMF tones. Users of all phones reported that when they access voicemail the VM app is not recognising DTMF tones. One clever user figured out that they DO work if you hold the key down for a certain amount of time, but getting it right is very difficult. Have a feeling that there is something that we have misconfigured, but can't figure it out. Any help appreciated. Cheers, Damian. FFF Managed Technology Ltd. 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Freeworlddialup
Hi there i just setup my asterisk to dial with freeworlddialup and i am trying to dial 411 voice xml service from freeworlddialup and I always get congestion/busy Is this normal Also dialing the tell number hangs up on me ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk-addon
I had the same problem. you did a CVS checkout on the latest version download this version instead it fixed the problem for me. http://www.asterisk.org/html/downloads/asterisk-sounds-1.0.7.tar.gz -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Altus Snyman Verzonden: woensdag 11 mei 2005 7:44 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] asterisk-addon Good day all I downloaded asterisk-addons to try and make asterisk log in the sql db but when I make a make install i get this error cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:162:77: macro AST_LIST_REMOVE passed 4 arguments, but takes just 3 app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:162: error: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:162: error: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:162: error: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 Please help ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: HINT
NOTE: Don't forget to reboot your phone after setting up asterisk it has to re subscribe for it to work!! -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Thorben Jensen Verzonden: zondag 8 mei 2005 21:03 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] Re: HINT Can you post a full dialplan example... I have a very complex dialplan but this is all you need. Put this line in your dial plan whenever you are making or recieving a call. exten = 201,hint,SIP/201 Also, will this only work for certain phones and atas also? I don't know. I use it for SNOM 190's ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk stat version 2 pdf output gives blank page
Hi there i installed the asterisk stat v2 but on the page i see an option for generating PDF files but when i click on the link it opens in a blank page, does anyone know how to fix this or do i need to install some additional package ? Thanks Sander Crombeen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mISDN error while compiling
Hi there all! Does anyone know what this error is??? I am trying to compile the mISDN in kernel 2.6.11.5 I get the same error in kernel 2.6.10.2 Someone?? HELP!!! WARNING: /lib/modules/2.6.11.5/kernel/drivers/isdn/hardware/mISDN/hfcmulti.ko needs unknown symbol pci_find_subsys ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel.conf multiple devices
Hi there my zaptel hardware is giving errors while loading but they seem to load just fine. the lights wil work and my wctdm card is also workin and the isdn works to But when I stop asterisk I have to reload al cards again is this normal? This is my zaptel.conf is there no way to group these because my te110p is giving an error that it cant find channel 35 but 35 belongs to my wctdm. Maybe my zaptel.conf is not that good, I cant find any documentation on multiple cards in one system Thanks ZT_SPANCONFIG failed on span 2: No such device or address (6) make: *** [loadlinux26] Error 1 ZT_CHANCONFIG failed on channel 35: No such device or address (6) FATAL: Error running install command for wcte11xp [EMAIL PROTECTED] src]# loadzone=nl defaultzone=nl span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=1,1,0,ccs,hdb3 bchan=4-18,20-34 # set this to 1-15,17-31 for E1 dchan=19 # set this to 16 for E1 fxoks=35-36 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] quadbri bristuff ztcfg fail
Please can anyone help me with my quadbri card I am desparate L I compiled the bristuff drivers and then I do -- Modprobe zaptel Insmod qozap.ko Ztcfg The it complains it cant find ZT_SPANCONFIG failed on span 1: No such device or address (6) --- When doing lsmod I can see qozap is loaded with zaptel but no entry in /proc/zaptel/ My zaptel.conf -- loadzone=nl defaultzone=nl span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] quadbri bristuff ztcfg fail
No udev installed on my system :( so that does not help me Thanks anyway -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Michael Bielicki Verzonden: vrijdag 29 april 2005 17:18 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] quadbri bristuff ztcfg fail smells like udev. Checkout README.udev in the zaptel directory. On 4/29/05, Sander [EMAIL PROTECTED] wrote: Please can anyone help me with my quadbri card I am desparate L I compiled the bristuff drivers and then I do -- Modprobe zaptel Insmod qozap.ko Ztcfg The it complains it can't find ZT_SPANCONFIG failed on span 1: No such device or address (6) --- When doing lsmod I can see qozap is loaded with zaptel but no entry in /proc/zaptel/ My zaptel.conf -- loadzone=nl defaultzone=nl span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michal Bielicki http://www.aefirion.org/ http://www.asterisk.com.pl/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] oh323 on @homeasterisk
Can you please detail the steps you have taken to successfully compile this on @home asterisk? Regards Mike - Original Message - From: CM Rahman Jr. [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, April 09, 2005 4:09 PM Subject: [Asterisk-Users] oh323 on @homeasterisk Anybody here added oh323 to @homeasterisk? I have compiled and add the oh323. I am wondering if anybody able to add the oh323 under web interface AMP? If anybody did it or know how to do it, please let me know. It has option for sip, IAX.. why not add h323 !! Thanks ** C.M. Rahman Jr. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.5 - Release Date: 7/04/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Set system time over the phone
No LAN what-so-ever. Customer is very paranoid.
Yes, sanitisation would be handy. Perhaps I should call an AGI file to do
this. Although I'm not sure how you can hack a system using only numbers
0-9, # and *. I'm sure there's a way!!!
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, April 05, 2005 8:02 PM
Subject: Re: [Asterisk-Users] Set system time over the phone
On Tue, Apr 05, 2005 at 09:45:54AM +1000, Mike Sander wrote:
I have installed Asterisk using the [EMAIL PROTECTED] image for a client that is
VoIP-a-phobic.
Hence the system cannot be connected to their LAN at all - don't ask why!
Does it have a lan connection at all? If so, you could use ntpd. Setting
the clock manually can have some side-effects and some services may
require running.
I have tested the clock at my installation lab, and all is fine, but they
might want to set/check it.
I know there is the SayUnixTime command, and it works fine to say the
time.
Is there a good dialplan command to test it? Best I've come across is
System, but this exits non-zero. Any ideas?
exten 456,1,Background(Please-set-time-mmddhhmm)
exten _.,1,System (date ${EXTEN})
Before passing input blindly to system, you need to sanitize it. E.g:
Any chance someone could dial a ';'? If so, that one can run an
arbitrary shell command (as Asterisk's user).
--
Tzafrir Cohen | New signature for new address and | VIM is
http://tzafrir.org.il | new homepage | a Mutt's
[EMAIL PROTECTED] || best
ICQ# 16849755 | Space reserved for other protocols | friend
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Re: [Asterisk-Users] Set system time over the phone
Looks good - thanks for the help!
Mike
- Original Message -
From: Roman Volf [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, April 05, 2005 4:48 PM
Subject: Re: [Asterisk-Users] Set system time over the phone
Another way is to do:
exten 456,1,Background(Please-set-time-mmddhhmm)
exten _.,1,System (echo ${EXTEN} /tmp/datetime )
Then have a cron job that runs every minute to check if file exists. For
example:
#!/bin/bash
if [ -f /tmp/datetime ] then
date `cat /tmp/datetime`
rm -f /tmp/datetime
fi
This should work fine.
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
Matt Riddell wrote:
Peter Bowyer wrote:
exten 456,1,Background(Please-set-time-mmddhhmm)
exten _.,1,System (date ${EXTEN})
If I dial 456 I get the message, so I type 04021305 (2nd April, 13:05).
On the console Asterisk reports the command Dial 04021305 exits
non-zero.
You need 'Read' instead of 'Background'.
No, because his next line is _.,1 so it will actually use the extension.
His problem is just one of permissions. Maybe he should use a suid prog
to set the date.
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[Asterisk-Users] Set system time over the phone
I have installed Asterisk using the [EMAIL PROTECTED] image for a client that is
VoIP-a-phobic.
Hence the system cannot be connected to their LAN at all - don't ask why!
I have tested the clock at my installation lab, and all is fine, but they
might want to set/check it.
I know there is the SayUnixTime command, and it works fine to say the time.
Is there a good dialplan command to test it? Best I've come across is
System, but this exits non-zero. Any ideas?
exten 456,1,Background(Please-set-time-mmddhhmm)
exten _.,1,System (date ${EXTEN})
If I dial 456 I get the message, so I type 04021305 (2nd April, 13:05).
On the console Asterisk reports the command Dial 04021305 exits non-zero.
If I then copy/paste into the shell, the command works.
Is there some weird brackets or something the System command is expecting
- the voip-info.org is up and down a lot at the mo.
Thanks
Mike
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[Asterisk-Users] Asterisk@Home H323
I am looking for a step-by-step on adding H323 to [EMAIL PROTECTED] So far I have installed [EMAIL PROTECTED], upgraded to the CVS-HEAD and followed instructions according to voip-info and this list's archives. I keep getting critical errors on compilation of H323, both Open 323 and OH323. Has anyone managed to install H323 with [EMAIL PROTECTED] If so, what steps did you perform. With Thanks Mike - Original Message - From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 29, 2005 11:41 AM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] 0.7 released Web Meetme is now installed by default and the meetme2 application is no longer needed. What does this mean exactly? Does this use the regular meetme as opposed to the meetme2 we had to setup before? On Mon, 28 Mar 2005 17:35:37 -0800 (PST), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: We had added a lot to this release to our one button install of Asterisk. Now you can have even more features automatically installed and configured. Asterisk 1.0.7 AMP 1-10-007 Flash Operator Panel 0.20 Redesigned WebMeetme weather agi scripts Midnight Commander We have added some of our most requested features. - Web Meetme is now installed by default and the meetme2 application is no longer needed. - we now have ZAP extension thanks to AMP 007 - weather.agi reads the current weather report using text to speech __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to do something random?
Logically, you should build something like this:
1. Pick a number between 1 and 3
2. Save the number to a variable indicating which line you are about to try
3. Check if it's free, if so make a call
4. If not, pick a number between 1 and 2
5. Make sure you haven't tried this number before (a loop and perhaps an
array of line numbers)
6. When you find a not-yet-tried number, check if it's free. If so, make a
call
7. If not, loop again to find the remaining number, check if free, if so
make a call.
8. If you get here, all lines are busy - play the busy tone.
I'm sorry my coding is not up to scratch, but this seems like a good
application for an AGI script as it can do arrays and looping easier, and
you could build this up to many lines.
Mike
Take a look at the Random() command.
MARK.
Ronald Wiplinger wrote:
I want to change the below lines:
exten = _011.,1,SetGroup(line1); set current group to
line
exten = _011.,2,CheckGroup(1); check line1 does
not have more than 1
exten = _011.,3,Dial,SIP/[EMAIL PROTECTED]; use line-1
exten = _011.,103,1,SetGroup(line2); set current
group to line
exten = _011.,104,CheckGroup(1); check line2 does not
have more than 1
exten = _011.,105,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}; use line-2
exten = _011.,205,1,SetGroup(line3); set current
group to line
exten = _011.,206,CheckGroup(1); check line3 does not
have more than 1
exten = _011.,207,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}; use line-3
exten = _011.,307,Busy; Play busy if all lines
already used
so that the three lines will be choosen random, but still only one
user per line.
Can you give me a hint?
BTW, I have not tested the lines yet, ... if you spot an error, please
point it out.
bye
Ronald
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Re: [Asterisk-Users] Flashpannel: How to get more than 28 buttons?
You can share them here: http://asterconf.hopto.org/ Mike - Original Message - From: Robert Rozman [EMAIL PROTECTED] To: Nicolás Gudiño [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 17, 2005 12:10 AM Subject: Re: [Asterisk-Users] Flashpannel: How to get more than 28 buttons? Hi, I'd also like to see alternative op_style.cfg. Can we establish some place to share them ? I've also one with smaller buttons (but will have to count them :-) ... Regards, Rob. - Original Message - From: Nicolás Gudiño [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 16, 2005 1:26 PM Subject: Re: [Asterisk-Users] Flashpannel: How to get more than 28 buttons? Hi Ronald, I have setup flash pannel, ... looks nice, but so far I could not configure it to get more than 4x7 buttons. I tried to make the buttons smaller, but than just the entire picture is smaller. What did you change in op_style.cfg? You can have literally hundred of buttons per screen, or multiple 'context' to split your buttons into several screens. I wll send you an alternate op_style.cfg with smaller buttons offlist. Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.3 - Release Date: 15/03/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Best DB
But Budwieser tastes like water to most Australian beer drinkers. (Now I'm in trouble!) Mike - Original Message - From: Chris Albertson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 18, 2005 11:48 AM Subject: RE: [Asterisk-Users] OT: Best DB What is the best truck? A recent survey finds that there are far more Ford Rangr pickup trucks on the road then there are Frightliner 18 wheelers In another survey we find that Chevy outnumbers Porche. Closer to home in the computer world, more people use MS Windows than Solaris. I think Budwieser outsells every other beer. In most organizations followers outnumber the leaders The poor will always outnumber the rich. Still interrested in that database poll? What's the best DB. First you must define best. After you do that the answer is easy. --- David Brodbeck [EMAIL PROTECTED] wrote: -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Top Deployed Databases poll shows following databases in use: SQL Server with 78%, Oracle - 55%, MySQL - 33% and PostgreSQL - 8%. I see they created this with Mysql, 78 + 55 + 44 + 8 = 185% I'm sure if you add in the others we would get to something around 300% deployment. Presumably some sites had more than one type of database in use. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.3 - Release Date: 15/03/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Help Site - cut down on Mailing List questions
This is a re-post as it was pointed out that I replied to a different thread instead of creating a new post. Sorry for the additional traffic. Mike Dear All, I understand the excitement surrounding a service like Asterisk, and how easy it is to jump in and ask a heap of questions. I also know how frustrating it can be dealing with a 200+ post per day mailing list as one of the question answerers. When I discovered Asterisk, I had a lot of study to do, because there are no real-world examples out there, just the trivial ones on the tiki and in the manual. I hope to propose a solution. I have (in a small time) downloaded and set up a repositor where we should all post our conf files, in an effort to get a big resource of a lot of different setups that we know just work. The program is simple, and looks like crap and is a testiment to my programming skills (or lack thereof). If anyone feels like re-coding or hosting this, let me know. You can find this at: asterconf.hopto.org (i think this has popups for the free DNS) or home.exetel.com.au/azyc/asterconf In the same philosophy as the GPL and wiki, it is open to all to search, view and download the conf code, however to post and add new categories, you must register. The site will not send you any mail or spam or anything. Of course, you should scrub your conf files for IP addresses and user/secrets, but otherwise, please post as much as you like. Please also include a description of the purpose of the post, and what type of service it runs on, for better searching. As a registered user, you are also free to add comments to other people's code snippets (but not change the code), and add more categories and sub-categories. I have started by creating categories for the most common conf files, under both working and broken sections. As a new (or old) asterisk user, if you are stuck, feel free to post your conf in the broken section and hopefully someone will come to help you. This will stop people posting codes to this list and flodding it. If we all use this resource, the we will reduce the amount of posts for people looking for instant setups, who don't want to use AMP or otherwise. That way, we can return this list to the discussion of Asterisk issues, rather than just a startup resource and helpdesk. I'm always interested in anyone's comments. Cheers Mike Sander sanderm at iprimus.com.au +61 2 401 010 289 (Australian mobile) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dock-n-talk connection to asterisk
Hi Peter. Look in last weeks (1/3/05) Sydney Morning Herald Tuesday IT liftout. They talk there about GSM gateways. It was made by Ericson I think, for around $1000. It's not meant for computer, rather as a FXO/FXS gateway to plug your house phone in for exactly the purpose you are talking about. Of course, if it is a FXO gateway, I'm sure a RJ cable (possibly crossover) will plug it in to a TD400 Digium card nicely to get what you want. I'm interested to know your progress, I have a few clients also interested in Sydney. Cheers Mike - Original Message - From: Peter Illmayer [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 05, 2005 2:06 PM Subject: [Asterisk-Users] Dock-n-talk connection to asterisk Hi ALL I'm looking for feedback on how well this unit integrates into asterisk via an ata. Is the audio quality any good as thats the first thing to upset the wife if its no good. I'm looking for a reasonably priced GSM gateway 1800mhz for use in Australia that works with an ata. Quite happy to import something that works well... Currently PSTN to mobile is $0.40c per minute and going to a selected provider, it will only cost $0.05c per minute so the savings are enormous for me, hence my interest in the DOck-n-Talk Any feedback would be very much appreciated ! -- Open WebMail Project (http://openwebmail.org) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.6.2 - Release Date: 4/03/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Help Site - cut down on Mailing List questions
Dear All, I understand the excitement surrounding a service like Asterisk, and how easy it is to jump in and ask a heap of questions. I also know how frustrating it can be dealing with a 200+ post per day mailing list as one of the question answerers. When I discovered Asterisk, I had a lot of study to do, because there are no real-world examples out there, just the trivial ones on the tiki and in the manual. I hope to propose a solution. I have (in a small time) downloaded and set up a repositor where we should all post our conf files, in an effort to get a big resource of a lot of different setups that we know just work. The program is simple, and looks like crap and is a testiment to my programming skills (or lack thereof). If anyone feels like re-coding or hosting this, let me know. You can find this at: asterconf.hopto.org (i think this has popups for the free DNS) or home.exetel.com.au/azyc/asterconf In the same philosophy as the GPL and wiki, it is open to all to search, view and download the conf code, however to post and add new categories, you must register. The site will not send you any mail or spam or anything. Of course, you should scrub your conf files for IP addresses and user/secrets, but otherwise, please post as much as you like. Please also include a description of the purpose of the post, and what type of service it runs on, for better searching. As a registered user, you are also free to add comments to other people's code snippets (but not change the code), and add more categories and sub-categories. I have started by creating categories for the most common conf files, under both working and broken sections. As a new (or old) asterisk user, if you are stuck, feel free to post your conf in the broken section and hopefully someone will come to help you. This will stop people posting codes to this list and flodding it. If we all use this resource, the we will reduce the amount of posts for people looking for instant setups, who don't want to use AMP or otherwise. That way, we can return this list to the discussion of Asterisk issues, rather than just a startup resource and helpdesk. I'm always interested in anyone's comments. Cheers Mike Sander [EMAIL PROTECTED] +61 2 401 010 289 (Australian mobile) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Manager API - multi Originate cal ls
I'm sure this has been said, but the [EMAIL PROTECTED] installation of Flash Operator Panel shows the handset shaking when a phone is ringing, so there is a way to do it. I'd search in there. Mike - Original Message - From: mattf [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 03, 2005 6:29 AM Subject: RE: [Asterisk-Users] Asterisk Manager API - multi Originate cal ls Well, I'm not sure about the current release as I have not tested this, but on older releases for RBS T1s you would get a manager event showing a RING state. As for PRI, SIP and IAX2 I'm not sure, this is an inconsistent feature that differes depending on what kind of trunk you are using and what network the person you are calling is on. The only sure thing you can tell is a call pickup in all cases, ringing is much harder to detect. MATT--- -Original Message- From: Thomas Miller [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 02, 2005 2:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Manager API - multi Originate cal ls Hi Matt, in your experience is there a 100% reliable way to know that the callee phone is ringing? In my situation I don't need to know if they pick up or not, I need to know (as reliably as possible) if the calee phone number is ringing. Thanks, Tom --- mattf [EMAIL PROTECTED] wrote: ActionID does not return in all events related to an Action sent, sometimes it will just send you a success message and nothing more. Just try Originating a call from a meetme room over an outside line. You will get about 150 lines of output and only one message will have the ActionID in it, the success message. On the other hand the callerID is placed on many more of the events in the output. It is still the case that if you do complex Manager Actions, the ONLY solution for tracking a call is to use a custom CallerID. Action: Originate Exten: 8600080 Channel: local/[EMAIL PROTECTED] Context: default Priority: 1 Callerid: DF345678901234567890 Actionid: AID45678901234567890 MATT--- -Original Message- From: Bill Seddon [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 02, 2005 8:06 AM To: Stephen Owen hosted; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Manager API - multi Originate calls read in places that you use originate command and wait for an event back, does that mean you cannot place another originate until the event comes back ? Not in my experience. Originate will not send an event to the caller until either the intended caller (that is the extension used in Originate) has picked up their phone or a timeout occurs because the intended caller does not pick up their phone. You can send as many originate requests as you like but they will fail if more than one uses the same extension at the same time. The issue you will face is determining which event generated by Asterisk belongs to which origination request. For this reason, the Manager API allows you to specify an ActionID on any command. An ActionID is any string of characters that you use to uniquely identify each command use issue. Asterisk will include the ActionID with each related event so you know which events to respond to and which to ignore. You will see many events generated by Asterisk only some of which will relate to your command. The others will be events that Asterisk raises (for example when a phone registers) or events in response to commands issues by other Manager API users and at the command line. Take a look at Nicolas Gudino's Flash Operator Panel ( www.asternic.org http://www.asternic.org/ ) as it used the manager API extensively (albeit through a proxy) and will typically make many requests via the Manger API. Is it true that multiple API connections to Asterisk Manager API will crash it (thinking of alternative way to crack the nut) Again, not in my experience. Lyquidity Solutions Limited +44 (0) 208 241 0500 _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Owen hosted Sent: March 02, 2005 12:28 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Manager API - multi Originate calls Been researching connecting over TCP\IP to Asterisk Manager API to initiate several concurrent calls to dial out. Prefer not to generate ASCII .call files. Question : I read in places that you use originate command and wait for an event back, does that mean you cannot place another originate until the event comes back ? Is it true that multiple API connections to Asterisk Manager API will crash it (thinking of alternative way to crack the nut) All help would be welcome - thanks Stephen Owen sip:[EMAIL PROTECTED] IM:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com
Re: [Asterisk-Users] Autodetecting faxes
That's all very well, but what do you do if you only have SIP extensions and IAX trunk - no Zaptel card. Will Fax detection still work at all? Thanks Mike - Original Message - From: Adrian Chapman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 08, 2005 8:24 PM Subject: Re: [Asterisk-Users] Autodetecting faxes Michael Welter wrote: Changing the order of things in extensions.conf around a smidge got it all working nicely :- [inbound-from-pstn] include = default exten = s,1,Answer exten = s,2,Wait,1 exten = s,3,Playback(thank-you-for-calling-please-wait-a-moment) exten = fax,1,Macro(faxreceive) exten = s,4,Do the normal phone call gubbins Is the position of the fax extension, between priorities 3 and 4, significant? What does 'show dialplan' display for the fax extension? It's there as much for flow readability as anything... The change of order was as much referring to moving the Playback forward from the voice handling macro, to give * time to hear the fax beep. Show Dialplan gives :- In each of my inbound call contexts 'fax' = 1. Macro(faxreceive) [pbx_config] No other mention at all. -- Adrian Chapman Director Trivas Ltd Business on the Move Mobility - Messaging - Infrastructure - Security - Remote Access 07796 690210 - 01582 626552 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.6 - Release Date: 7/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to download CVS with attended transfers
Hi I know that attended transfers are only available in the CVS Head. I downloaded the asterisk-update.sh script from voip-info.com and ran it with these parameters ./asterisk-update.sh update dev It looked as tho CVS HEAD was downloading and compiling, although it couldn't download the addons. However, now it's up and running, only blind transfers work with #, and I cannot change the blind transfer key to ##, it only takes the first character. And Attended transfers still isn't running. Is there something I've missed. The version info reports: Asterisk CVS-v1-0-02/03/05-10:24:22 Any help would be great. Thanks Mike -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.4 - Release Date: 1/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to download CVS with attended transfers
Hi I know that attended transfers are only available in the CVS Head. I downloaded the asterisk-update.sh script from voip-info.com and ran it with these parameters ./asterisk-update.sh update dev It looked as tho CVS HEAD was downloading and compiling, although it couldn't download the addons. However, now it's up and running, only blind transfers work with #, and I cannot change the blind transfer key to ##, it only takes the first character. And Attended transfers still isn't running. Is there something I've missed? The version info reports: Asterisk CVS-v1-0-02/03/05-10:24:22 Any help would be great. Thanks Mike -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.4 - Release Date: 1/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multi Asterisk Server Transfers
I believe this is what I have, but it still insists on running the transfer from the head office. Example: Provider --- IAX --- Head Office Provider --- SIP --- Remote Office Provider --- PSTN (Provider is the same * server in all cases) Call comes from PSTN to Head office. Head office transfers to 0 where is SIP extension according to Provider and 0 is to dial out on the trunk. Call is then connected as follows. PSTN - Provider - Head Office - Provider - Remote But after it is transferred, I want the resulting route to be: PSTN - Provider - Remote Otherwise Head office has 2 times the bandwidth running through it for a call not even going to one of it's own extensions. I had throught that the IAX connection between Provider and Head Office would pass off calls that way. Let me know, but thanks for all the help so far. Mike Instead I'd go for a co-located Asterisk that the remote SIP devices register with, and then link both * boxes (co-located and central office) using IAX2 with IAX native transfers enabled. Of course this means that the office * _only_ talks IAX and that all calls to the remote SIP clients _always_ go thru the co-located box (with its extra bandwidth). SER certainly is another way to go (as mentioned before), but in this specific setup I assume it complicates matters unnecessarily. Cheers, Philipp -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.1 - Release Date: 27/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multi Asterisk Server Transfers
Simple as that? Anyone know a good IAX phone (not softphone)? Thanks Mike Then you need to use the same protocol to the provider. One office is using SIP, the other is using IAX. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.1 - Release Date: 27/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multi Asterisk Server Transfers
Hi, We are in the business of setting up * servers for businesses, attached via IAX trunks to our VoIP provider (also using *). I have a client with a head office * server, who wants a number of remote offices, with just 1 SIP connection to each. I can arrange this no probs with our providers, but there are issues with transfer. I don't want the remote offices making their direct SIP connection to the head office, because bandwidth is limited and then for them to make an outgoing call, the head office has both an incoming and outgoing connection - or double the bandwidth. This is the same for an incoming call to head office that gets transferred to the remote, the call stays with the head office * server, and the server makes another outgoing call to the remote office. All these calls are free, but use double the bandwidth. The question: The remote offices can make direct SIP connections to our provider. If the head office * transfers a call, then the server releases the call entirely back to the providers * server and calls from there. I.E. call in to head office from PSTN through the provider. Call gets transferred to the remote office. Head office could then unplug/burn/blowup their asterisk server without disrupting the call between the remote office and the PSTN network. Is this possible? Companies with multiple * servers in many remote office, surely have this system, to conserve bandwidth? How is the transfer made? Mostly we are using X-PRO systems/Grandstream, with the [EMAIL PROTECTED] basic release. Thanks Mike -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.5 - Release Date: 26/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multi Asterisk Server Transfers
I agree with you. If every office had a * server, it would be fine. i.e. Office 1 rings office 2, then gets transferred to office 3, then connection is direct from office 1 to 3, and 2 releases all contact. However, what if office 3 is a 1 person office, with just a single SIP phone connected to the VoIP provider. Full IAX trunking can do hand-offs quite simply, I think, but when the destination is a single SIP connection, things get messy. Is this relevant to your answer, because I'm a little confused now? With thanks Mike Seems strange to be handling multiple * servers over SIP and ignoring IAX2. I'd be inclined to trunk between offices over IAX2. In fact, I'd use and IAX2 based ITSP and then be able to hand off calls in a .reinvite fashion without all the messy port handling. In addition you save on bandwidth by trunking multiple calls over one IAX2 connection. Less IP overhead, between offices and to the ITSP. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.5 - Release Date: 26/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SetGroup and CheckGroup problems
I have a rather long dial plan, but it includes support for call waiting.
However, the setgroup checkgroup commands don't seem to be working. Can
anyone help on this one?
Excerpts are below. First exten-vm is dialed and then dial-new.
As I understand, priority 1 increments the active channels for the caller
and then in dial-new priority 8 increments for Arg3, or the Callee
extension. Problem is, that priority 9 always goes on to 10 (i.e. group
never is on-the-phone.
Am I missing something?
When ext201 dials 202, CLI shows:
-- Executing Macro(SIP/201-8571, exten-vm|202|202) in new stack
-- Executing SetGroup(SIP/201-8571, 201) in new stack
-- Executing SetMusicOnHold(SIP/201-8571, default) in new stack
-- Executing SetVar(SIP/201-8571, FROMCONTEXT=exten-vm) in new stack
-- Executing GotoIf(SIP/201-8571, 0?9:5) in new stack
-- Goto (macro-exten-vm,s,5)
-- Executing Macro(SIP/201-8571, dial-new|15|tr|202|202) in new
stack
-- Executing DBget(SIP/201-8571, CallForwardIm=CF/202) in new stack
-- DBget: varname=CallForwardIm, family=CF, key=202
-- DBget: Value not found in database.
-- Executing Goto(SIP/201-8571, s|4) in new stack
-- Goto (macro-dial-new,s,4)
-- Executing DBget(SIP/201-8571, DNDStatus=DND/202) in new stack
-- DBget: varname=DNDStatus, family=DND, key=202
-- DBget: Value not found in database.
-- Executing Goto(SIP/201-8571, s|8) in new stack
-- Goto (macro-dial-new,s,8)
-- Executing SetGroup(SIP/201-8571, 202) in new stack
I'll be most grateful for any assistance.
Thanks
Mike
[macro-exten-vm]
exten = s,1,SetGroup(${CALLERIDNUM})
exten = s,2,SetMusicOnHold(default)
exten = s,3,Setvar(FROMCONTEXT=exten-vm) exten =
s,4,GotoIf($[${CALLERIDNUM} = ${ARG2}]?9:5) ;check self-voicemail exten =
s,5,Macro(dial-new,${RINGTIMER},${DIAL_OPTIONS},${ARG2},${ARG1})
[macro-dial-new]
;now check if destination is on a call
exten = s,8,SetGroup(${ARG3})
exten = s,9,CheckGroup(1)
;go to 110 then 25 if on the phone (CW handler), go to 10 if not on the
phone
exten = s,110,Goto(s,25)
;line is clear, begin dial sequence
exten = s,10,Setvar(ChanType=${E${ARG3}}) ;Get the channel type
exten = s,11,Dial(${ChanType}/${ARG3},${ARG1},${ARG2})
Mike Sander
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[Asterisk-Users] SetGroup, CheckGroup
I have a rather long dial plan, but it includes support for call waiting.
However, the setgroup checkgroup commands don't seem to be working. Can
anyone help on this one?
Excerpts are below. First exten-vm is dialed and then dial-new.
As I understand, priority 1 increments the active channels for the caller
and then in dial-new priority 8 increments for Arg3, or the Callee
extension. Problem is, that priority 9 always goes on to 10 (i.e. group
never is on-the-phone.
Am I missing something?
When ext201 dials 202, CLI shows:
-- Executing Macro(SIP/201-8571, exten-vm|202|202) in new stack
-- Executing SetGroup(SIP/201-8571, 201) in new stack
-- Executing SetMusicOnHold(SIP/201-8571, default) in new stack
-- Executing SetVar(SIP/201-8571, FROMCONTEXT=exten-vm) in new stack
-- Executing GotoIf(SIP/201-8571, 0?9:5) in new stack
-- Goto (macro-exten-vm,s,5)
-- Executing Macro(SIP/201-8571, dial-new|15|tr|202|202) in new
stack
-- Executing DBget(SIP/201-8571, CallForwardIm=CF/202) in new stack
-- DBget: varname=CallForwardIm, family=CF, key=202
-- DBget: Value not found in database.
-- Executing Goto(SIP/201-8571, s|4) in new stack
-- Goto (macro-dial-new,s,4)
-- Executing DBget(SIP/201-8571, DNDStatus=DND/202) in new stack
-- DBget: varname=DNDStatus, family=DND, key=202
-- DBget: Value not found in database.
-- Executing Goto(SIP/201-8571, s|8) in new stack
-- Goto (macro-dial-new,s,8)
-- Executing SetGroup(SIP/201-8571, 202) in new stack
I'll be most grateful for any assistance.
Thanks
Mike
[macro-exten-vm]
exten = s,1,SetGroup(${CALLERIDNUM})
exten = s,2,SetMusicOnHold(default)
exten = s,3,Setvar(FROMCONTEXT=exten-vm)
exten = s,4,GotoIf($[${CALLERIDNUM} = ${ARG2}]?9:5) ;check self-voicemail
exten = s,5,Macro(dial-new,${RINGTIMER},${DIAL_OPTIONS},${ARG2},${ARG1})
[macro-dial-new]
;now check if destination is on a call
exten = s,8,SetGroup(${ARG3})
exten = s,9,CheckGroup(1)
;go to 110 then 25 if on the phone (CW handler), go to 10 if not on the
phone
exten = s,110,Goto(s,25)
;line is clear, begin dial sequence
exten = s,10,Setvar(ChanType=${E${ARG3}}) ;Get the channel type
exten = s,11,Dial(${ChanType}/${ARG3},${ARG1},${ARG2})
Mike Sander
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RE: [Asterisk-Users] Call Pickup
I have a rather long dial plan, but it includes support for call waiting.
However, the setgroup checkgroup commands don't seem to be working. Can
anyone help on this one?
Excerpts are below. First exten-vm is dialed and then dial-new.
As I understand, priority 1 increments the active channels for the caller
and then in dial-new priority 8 increments for Arg3, or the Callee
extension. Problem is, that priority 9 always goes on to 10 (i.e. group
never is on-the-phone.
Am I missing something?
When ext201 dials 202, CLI shows:
-- Executing Macro(SIP/201-8571, exten-vm|202|202) in new stack
-- Executing SetGroup(SIP/201-8571, 201) in new stack
-- Executing SetMusicOnHold(SIP/201-8571, default) in new stack
-- Executing SetVar(SIP/201-8571, FROMCONTEXT=exten-vm) in new stack
-- Executing GotoIf(SIP/201-8571, 0?9:5) in new stack
-- Goto (macro-exten-vm,s,5)
-- Executing Macro(SIP/201-8571, dial-new|15|tr|202|202) in new
stack
-- Executing DBget(SIP/201-8571, CallForwardIm=CF/202) in new stack
-- DBget: varname=CallForwardIm, family=CF, key=202
-- DBget: Value not found in database.
-- Executing Goto(SIP/201-8571, s|4) in new stack
-- Goto (macro-dial-new,s,4)
-- Executing DBget(SIP/201-8571, DNDStatus=DND/202) in new stack
-- DBget: varname=DNDStatus, family=DND, key=202
-- DBget: Value not found in database.
-- Executing Goto(SIP/201-8571, s|8) in new stack
-- Goto (macro-dial-new,s,8)
-- Executing SetGroup(SIP/201-8571, 202) in new stack
I'll be most grateful for any assistance.
Thanks
Mike
[macro-exten-vm]
exten = s,1,SetGroup(${CALLERIDNUM})
exten = s,2,SetMusicOnHold(default)
exten = s,3,Setvar(FROMCONTEXT=exten-vm) exten =
s,4,GotoIf($[${CALLERIDNUM} = ${ARG2}]?9:5) ;check self-voicemail exten =
s,5,Macro(dial-new,${RINGTIMER},${DIAL_OPTIONS},${ARG2},${ARG1})
[macro-dial-new]
;now check if destination is on a call
exten = s,8,SetGroup(${ARG3})
exten = s,9,CheckGroup(1)
;go to 110 then 25 if on the phone (CW handler), go to 10 if not on the
phone
exten = s,110,Goto(s,25)
;line is clear, begin dial sequence
exten = s,10,Setvar(ChanType=${E${ARG3}}) ;Get the channel type
exten = s,11,Dial(${ChanType}/${ARG3},${ARG1},${ARG2})
Mike Sander
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RE: [Asterisk-Users] SetGroup and CheckGroup problems
Excuse my continued denseness, but I'm still not getting the groups concept.
I have 1 IAX trunk allowing multiple incoming and outgoing calls, and about
10 SIP channels. I don't have any ZAP cards or channels configured. Is the
SetGroup command type intended mainly for Zaptel interfaces??
I changed the CheckGroup(1) command to GetGroupCount(${ARG3}) where ARG3 is
the extension being dialed. Then I added SetVar(GPCNT=${GROUPCOUNT}) so I
could see the value of GROUPCOUNT in the CLI debug.
The answer is usually 1, whether the destination is on a call or not. When
they are conferencing with 2 external calls, it shows 2, but there doesn't
seem to by rhyme or reason. It makes sense to me to show 1 when they are on
the call, 2 when they have 2 going etc, but if they aren't on any calls, it
should show 0.
Am I missing something here, I'm sure it's really obvious.
With thanks
Mike
Mike Sander wrote:
I have a rather long dial plan, but it includes support for call waiting.
However, the setgroup checkgroup commands don't seem to be working. Can
anyone help on this one?
Long story short: you cannot put a channel into two groups, unless you
add categories to your group names. Calling SetGroup multiple times
without category designators just replaces the channel's group each time
you call it.
However, you do not need to use SetGroup/CheckGroup to check a group's
status; you can use GetGroupCount to directly check any group you want,
even one that the current channel is not in. If you are using CVS HEAD,
you can even use GetGroupMatchCount to get counts of multiple groups
with similar names. Also in CVS HEAD, you can set OUTBOUND_GROUP before
calling Dial(), and the channels it creates to actually call the targets
will be automatically placed into that group, as if you had used
SetGroup on them (which you cannot do normally, since you can't run any
dialplan code on those channels).
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Version: 7.0.300 / Virus Database: 265.7.3 - Release Date: 24/01/2005
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RE: [Asterisk-Users] call return?
For me this worked straight out of the box with [EMAIL PROTECTED] 0.3 Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry Hills N.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Polk Sent: Sunday, 23 January 2005 4:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] call return? Hi: Can any one point me in the rite direction on this? I am using asterisk at home for learning purposes. I am trying to get the triditional *69 working. Has there been any success in getting it to announce the number and get it to give you the option to call back? Chris - Original Message - From: Diego Ventrice [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, January 22, 2005 8:03 AM Subject: Re: [Asterisk-Users] softswitch dilemma Thanks for answering Chad, Actually, I just want to Switch traffic between wholesale providers (my customers) which actually terminate traffic (or not, some of them have just controllers-softswitches like the one Im willing to set up) collect CDRs and bill them =) I have no gateways of my own (of any kind) so Im not originating nor terminating calls, just switching traffic is my goal, all this people use h.323 of course. Any advice would be appreciated. Thanks for your help D. Date: Fri, 21 Jan 2005 22:23:58 -0600 (CST) From: Chad Whitten [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] softswitch dilemma To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 are you looking to do actual pstn to voip termination? if so, then you are gonna need ss7, cama and imt trunks - things which asterisk doesnt necessarily support. now if you just want to buy pri/t1 from the local telco and sell voip services off an asterisk server that gets back to the pstn over these pri/t1's, then yes, asterisk can do this. Diego Ventrice said: Hello everybody, Im new to the list and also new to asterisk, Im wondering if I could set up asterisk as a softswitch, I guess for what I've been reading that It could be possible but almost all the info and documentation Ive found so far is about asterisk as a PBX, etc. Im willing to set a small voip wholesale traffic bussiness and Im not quite sure asterisk is the right chocie for that. An asterisk-ser or an asterisk-vocal combination may be the answer ? Thanks in advance for any help. Diego -- Chad Whitten ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UPS for Asterisk
I'd considering an UPS backup system for my Asterisk server. I understand this is a linux issue, not a * issue, except for the following... Is the harddisk activity on a standard asterisk install such that I don't really have to worry if the power cuts?? As I understand, if HD activity is minimal, the probability of HD failure is significantly reduced. P.S. Power regulation is not needed, only protection against instantaneous power loss. Mike Sander -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems transferring calls - HELP!
Hi. I have the following weird phenomenon on a standard * @ home installation. I use X-PRO on 2 or 3 computers. Blind Transfers: With incoming calls: I click transfer then dial extension the click transfer again. Result: Hangs up on incoming caller. With outgoing calls: I click transfer then dial extension the click transfer again. Result: Transfer is ok, dialed extension connects to outgoing call. Assisted Transfers: Place incoming caller on hold. Get new line and dial extension. Chat with extension then click transfer and the line number of incoming caller. Result: Incoming caller can hear new extension ok, but new extension can only hear music. OR Place incoming caller on hold. Get new line and dial extension. Chat with extension then place him on hold too. Return to incoming caller, click transfer and the line number of dialed extension. Result: Incoming caller can hear music only, but new extension can incoming caller. This is very perplexing. It is like the XPro is interacting with * in a way that it is transferring the MOH channel, not the person. And the order is weird. Transferring 1 to 2 gives reverse result to transferring 2 to 1. When I had individual SIP accounts to our VoIP provider's * server, rather than our own, everything worked fine. Please help if you can, this is baking my noodle!!! Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry Hills N.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.1 - Release Date: 19/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems transferring calls - Part 2!
Ok. I've done more research and testing and here are the details. It is using the dialparties.agi (http://www.sprackett.com/asterisk/) file to dial . Originally with the dial options tr. I changed the options to Tt but no change. Transferring internal extensions between each other works fine. Example: 201 calls 202. 201 transfers 202 to 203. Transferring the IAX trunk to other internal is weird, as per my previous email. Example: DID calls 201. 201 transfers DID to 202. DID is either hungup or half connected (DID gets connected to 202, but 202 only hears music. DID can hear 202, even though 202 is hearing music). At the moment I can only transfer trunk calls through the parking system, which is a pain to teach people about... I'm really stumped on this one. Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry Hills N.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.1 - Release Date: 19/01/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.1 - Release Date: 19/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DIDs anywhere but here?
We have DID's in 5 Australian cities for $5 per month. Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry Hills N.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Tuesday, 18 January 2005 3:46 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] DIDs anywhere but here? Are there affordable DIDs (preferably IAX) available anywhere outside the US? I want to use it to meet ICANN requirements for providing a valid phone number, yet pre-empting some of the telemarketing calls my domain registrations generate. (Yes, I asked a similar question about 900# availability before). I'd prefer to have a number somewhere outside the NANP, preferably an asian country. This number will (obviously) be low-volume (minutes/month at the most), and shouldn't cost more than a couple of bucks. Maybe a list member knows and/or is using one? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.0 - Release Date: 17/01/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.0 - Release Date: 17/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transferring calls on Asterisk with X-Lite
I am having trouble transferring calls using asterisk. I think it is my * installation, because this worked fine with the same system when it was hosted at our VoIP providers. I receive a call on my IAX Trunk, to my extensions. I speak to the incoming call and tell them Ill just transfer the call. I click Transfer, dial extension and click Transfer again. Normally the call will disappear on my system and start ringing on the new extension. In this case, the call just hangs up. Any Ideas? Do you have to setup transfers in the extensions at all? Does this have something to do with the Reinvite status of the SIP phones? With Thanks Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry HillsN.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.0 - Release Date: 17/01/2005 image001.jpg___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] On Hold music
Do you have Zaptel cards installed? You need to have a timer installed (whatever that means). If you dont have a zaptel card, then use ztdummy to fake one. You need to download and compile the zaptel drivers (from asterisk website). Edit the makefile and find the line: TZOBJS=zonedata.lo tonezone.lo LIBTONEZONE=libtonezone.so.1.0 MODULES=zaptel tor2 torisa wcusb wcfxo wcfxs \ ztdynamic ztd-eth wct1xxp wct4xxp wcte11xp # ztdummy #MODULES+=wcfxsusb Then remove the # before ztdummy Type Make All Type Make Install Add a line to load the module ztdummy on boot using the /etc/rc.d files The command is modprobe ztdummy More information at: http://www.voip-info.org/tiki-print.php?page=Asterisk+timer+ztdummy Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry Hills N.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Computer Onsite Support Sent: Tuesday, 18 January 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] On Hold music Can anyone of you help me out with this issue. My Asterisk is working fine except my music-on-hold will NOT work even though I just retry three different other machines with different board and sound. [Manny Teixeira] al Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Manjit Riat Sent: Monday, January 17, 2005 8:42 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP URL for incoming I want to set up my asterisk to receive SIP calls using the URL [EMAIL PROTECTED] . I have my own DNS server but would like know what entry goes into it as I have never set up SRV records before. (if it matter it is a BIND dns server). thanx -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.0 - Release Date: 17/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk@Home systems
I am having trouble setting up Meetme with this CD. I have the latest which
was posted on sourceforge about 2-3 days ago. It seems to come with meetme
8200 and 8201 rooms, but I am getting invalid messages.
Can anyone help.
The Meetme.conf is:
conf = 8200
conf = 8201
The extensions are:
exten = _8XXX,1,Answer
exten = _8XXX,2,Wait(1)
exten = _8XXX,3,GotoIf($[${CALLERIDNUM} = ${EXTEN:1}]?5:4)
exten = _8XXX,4,MeetMe(${EXTEN}|sM)
exten = _8XXX,5,MeetMe(${EXTEN}|asM)
The extensions set up are 200 and 201.
I assume you dial 8200 to be administrator of your own meetme room.
Help if anyone knows please.
Thanks
Mike
--
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Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.6.11 - Release Date: 12/01/2005
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[Asterisk-Users] Limit outgoing trunk calls
Hi, We have an IAX provider that limits incoming IAX trunk calls, based on how many lines you purchase, but gives unlimited outgoing calls. I want to use the local Asterisk server to limit the outgoing number of calls, to retain high bandwidth. I.E. If we can only support 10 symultaneous high-quality calls on our broadband connection, I want the 11th person that dials the outgoing line extension to get a congestion/busy signal. Does Asterisk have a way of tracking how many people are on the trunk at one time, and accept/reject new calls based on that? Is it though the dialplan or the iax.conf? Thanks Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry HillsN.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.11 - Release Date: 12/01/2005 image001.jpg___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_capi version 0.1.0 released
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Liaan van der Merwe Sent: Thursday, March 13, 2003 1:20 PM I just installed all the latest isdn4linux stuff.. ^^ You need capi4linux, not isdn4linux (although it is part of the isdn4linux CVS store). Still same error Maybe asterisk wont work with external isdn devices.. any ideas? What device are you trying to use? A bit more info would be helpfull. Sander ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
