Re: [asterisk-users] Belgian BRI (euroisdn): what to use for a B410P
On 05/10/12 00:09, Richard Mudgett wrote: Please just reply to the mailing list. oops, that was my intention, my bad. Are you able to make calls when in PTP mode? I just tested: yes it seams so! The warning message is just complaining about receiving unexpected TEI management messages because the span is in PTP mode. It is otherwise benign if the line is really PTP. If you can make calls, please create a JIRA issue on the PRI project so the message level can be reduced. Please attach an intense pri debug output showing the received MDL messages. pri set debug 2 span 4 https://issues.asterisk.org/jira Richard Will do. I suppose there is no way to make them disappear already, except for turning of WARNING messages. BC -- _ -- 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] looking for solid state like PC suitable for Asterisk
Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? thx!! BC -- _ -- 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] where can i find code documentation
Its rather surprising that i'm unable to find the code documentation generated by make progdocs. It should be /usr/share or /usr/local/share but it does not appear to be there. Any clue? -- -aft -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium IP Phones
Hello, I've been using the Digium D40's for a few weeks now and I think they are good for the price. There are a few UI problems but I hope/expect they will be resolved in a firmware update or two. Haven't looked at the SDK yet. Thanks, Dennis On Thu, May 10, 2012 at 2:38 AM, Danny Dias ing.diasda...@gmail.com wrote: Hello, Im looking to buy a digium phone D70 unit just for testing on lab; to really understand the phone and features. I cant find any website with opinions; any here? Are they really valuable to the price? (D70 quite expensive) Does the SDK for building apps is usable? Can you build powerfull apps? Examples? Many thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium IP Phones
We've just had one of each delivered for us to play with in our lab (Literally an hour ago!). Not had chance to play with them yet, But initial thoughts are they look good. Build quality seems fine for the price. I'll form more of an opinion when i get chance to play with them properly tomorrow. I don't think the SDK is available yet (I've not been able to find it on the digium site). I'm itching to get my hands on it though! My first thought when seeing the D70 and looking at the screen for the speed dial keys was I hope we can use this screen in for the apps, It's perfect for a tetris clone. :) Cheers, AJ. - Original Message - From: Danny Dias ing.diasda...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, 10 May, 2012 2:38:02 AM Subject: [asterisk-users] Digium IP Phones Hello, Im looking to buy a digium phone D70 unit just for testing on lab; to really understand the phone and features. I cant find any website with opinions; any here? Are they really valuable to the price? (D70 quite expensive) Does the SDK for building apps is usable? Can you build powerfull apps? Examples? Many thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium IP Phones
On 05/09/2012 08:38 PM, Danny Dias wrote: Hello, Im looking to buy a digium phone D70 unit just for testing on lab; to really understand the phone and features. I cant find any website with opinions; any here? Are they really valuable to the price? (D70 quite expensive) Does the SDK for building apps is usable? Can you build powerfull apps? Examples? The phone app SDK has not been released yet, it's still under development. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] where can i find code documentation
On 05/10/2012 05:08 AM, Arif Hossain wrote: Its rather surprising that i'm unable to find the code documentation generated by make progdocs. It should be /usr/share or /usr/local/share but it does not appear to be there. Any clue? It is generated in the 'doc' directory of the source code tree. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Increasing voice volume without getting echo or entered digit problem
Dears; How I can increase the voice volume in the analogue line (at dahdi port) without getting a problems in the entered digits (when using Background function)? I got to know previously that it is possible to increase the gain at hardware and not software, this is better to avoid getting a problems. But where? I am using DAHDI 2.4 and another machine has DAHDI 2.6 Using txgain and rxgain from chan_dahdi.conf will not help, it will increase the voice volume but with the following problems: 1) Suddenly the call will be disconnected while we are talking. 2) When calling the Asterisk box and we entered the digits, it is failing to collect it (sometime does not collect it correctly and sometime it collects the digit duplicated). 3) Echo problem. So I need to know how to increase the voice volume from another place? Appreciate the kindly help. Regards Bilal -- _ -- 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] looking for solid state like PC suitable for Asterisk
I use HP Thin Clients with AstLinux installed. HP 5720's are available on eBay for not much money, or there are many small boards available new if you don't or can't use used. 10 watts, no fan, no HD Not sure what might be available in your part of the world, but there are Sockris and ALIX flash based boards. AstLinux has special configurations for these. I have 20-30 AstLinux on thin clients working without a belch on a private collectors network John Novack Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? thx!! BC -- _ -- 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 -- 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
Re: [asterisk-users] looking for solid state like PC suitable for Asterisk
Another option is to get those routers that are capable of running dd-wrt firmware with USB ports(for storage) This option is rather good if you don't need any VoIP cards and if you are OK to use sip/iax2 etc trunks. I have my wifi router with dd-wrt firmware running asterisk for home use. It's cheap, small, uses less power, noiseless and :) just cool Sent from my iPhone On May 10, 2012, at 7:35 PM, John Novack jnov...@stromberg-carlson.org wrote: I use HP Thin Clients with AstLinux installed. HP 5720's are available on eBay for not much money, or there are many small boards available new if you don't or can't use used. 10 watts, no fan, no HD Not sure what might be available in your part of the world, but there are Sockris and ALIX flash based boards. AstLinux has special configurations for these. I have 20-30 AstLinux on thin clients working without a belch on a private collectors network John Novack Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? thx!! BC -- _ -- 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 -- 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
Re: [asterisk-users] looking for solid state like PC suitable for Asterisk
On Thursday 10 May 2012, Bart Coninckx wrote: I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? Raspberry Pi would be the obvious choice, surely? -- AJS Answers come *after* questions. -- _ -- 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] Community event: Open Source Realtime Dinner in Barcelona - June 13th
Hello! I will be running an Asterisk SIP Masterclass - the last one - in Barcelona in June. During this week, I will organize a dinner for everyone working with or interested in Asterisk, Kamailio and other Open Source platforms for realtime communication. It's June 13th somewhere in Barcelona - location will be announced later. You pay our own dinner (unless we can find sponsors) and enjoy the geeky company for free! To join the event, use this Facebook event https://www.facebook.com/events/307548349321608/ See you in Barcelona! /O -- http://edvina.net - Open Unified Communication - training, consulting, workshops -- _ -- 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] looking for solid state like PC suitable for Asterisk
On 05/10/12 13:49, A J Stiles wrote: On Thursday 10 May 2012, Bart Coninckx wrote: I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? Raspberry Pi would be the obvious choice, surely? I'm in the waiting queue for one, but they still seem to be needing to sell one per person, while I need many. Not a bad idea though, BC -- _ -- 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] looking for solid state like PC suitable for Asterisk
That's Soekris I suppose. Never heard of them, but it looks mighty interesting. Cheers, BC On 05/10/12 13:35, John Novack wrote: I use HP Thin Clients with AstLinux installed. HP 5720's are available on eBay for not much money, or there are many small boards available new if you don't or can't use used. 10 watts, no fan, no HD Not sure what might be available in your part of the world, but there are Sockris and ALIX flash based boards. AstLinux has special configurations for these. I have 20-30 AstLinux on thin clients working without a belch on a private collectors network John Novack Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? thx!! BC -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for solid state like PC suitable for Asterisk
This is for an ISDN project, but Beronet has ISDN gateways with ethernet, so even that might not be an issue, cheers, BC On 05/10/12 13:43, Arstan Jusupov wrote: Another option is to get those routers that are capable of running dd-wrt firmware with USB ports(for storage) This option is rather good if you don't need any VoIP cards and if you are OK to use sip/iax2 etc trunks. I have my wifi router with dd-wrt firmware running asterisk for home use. It's cheap, small, uses less power, noiseless and :) just cool Sent from my iPhone On May 10, 2012, at 7:35 PM, John Novackjnov...@stromberg-carlson.org wrote: I use HP Thin Clients with AstLinux installed. HP 5720's are available on eBay for not much money, or there are many small boards available new if you don't or can't use used. 10 watts, no fan, no HD Not sure what might be available in your part of the world, but there are Sockris and ALIX flash based boards. AstLinux has special configurations for these. I have 20-30 AstLinux on thin clients working without a belch on a private collectors network John Novack Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? thx!! BC -- _ -- 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 -- 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
Re: [asterisk-users] looking for solid state like PC suitable for Asterisk
On Thu, 2012-05-10 at 12:49 +0100, A J Stiles wrote: On Thursday 10 May 2012, Bart Coninckx wrote: I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? Raspberry Pi would be the obvious choice, surely? I'll be trying that the moment mine arrives :) -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] looking for solid state like PC suitable for Asterisk
Correct. I have never been accused of being a good speller! JN Bart Coninckx wrote: That's Soekris I suppose. Never heard of them, but it looks mighty interesting. Cheers, BC On 05/10/12 13:35, John Novack wrote: I use HP Thin Clients with AstLinux installed. HP 5720's are available on eBay for not much money, or there are many small boards available new if you don't or can't use used. 10 watts, no fan, no HD Not sure what might be available in your part of the world, but there are Sockris and ALIX flash based boards. AstLinux has special configurations for these. I have 20-30 AstLinux on thin clients working without a belch on a private collectors network John Novack Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? thx!! BC -- _ -- 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 -- 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
Re: [asterisk-users] looking for solid state like PC suitable for Asterisk
This thread may interest you. Add a SSD and RAM and you're good to go! http://pbxinaflash.com/community/index.php?threads/diy-piaf2-server-200. 12460/ -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Thursday, May 10, 2012 8:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] looking for solid state like PC suitable for Asterisk Correct. I have never been accused of being a good speller! JN Bart Coninckx wrote: That's Soekris I suppose. Never heard of them, but it looks mighty interesting. Cheers, BC On 05/10/12 13:35, John Novack wrote: I use HP Thin Clients with AstLinux installed. HP 5720's are available on eBay for not much money, or there are many small boards available new if you don't or can't use used. 10 watts, no fan, no HD Not sure what might be available in your part of the world, but there are Sockris and ALIX flash based boards. AstLinux has special configurations for these. I have 20-30 AstLinux on thin clients working without a belch on a private collectors network John Novack Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? thx!! BC -- _ -- 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 -- 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
Re: [asterisk-users] looking for solid state like PC suitable for Asterisk
Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? Have a look at the Blackbochs SBC***. It is small, low power, plenty of storage options, built in analog telephony ports, etc: http://www.rockbochs.com/products/blackbochs-sbc --Tim ***Yes, I'm affiliated with the product/company, but it is on topic for this discussion. My apologies if this offends anyone. -- _ -- 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] looking for solid state like PC suitable for Asterisk
- Original Message - On Thursday 10 May 2012, Bart Coninckx wrote: I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? Raspberry Pi would be the obvious choice, surely? The hype around the Raspberry Pi is enormous. I would not consider it a real option for production voice until it's had a chance to mature and be available for some time to iron out the bugs, both hardware and software related. My $0.02 USD. --Tim -- _ -- 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] looking for solid state like PC suitable for Asterisk
On Thu, 10 May 2012, A J Stiles wrote: On Thursday 10 May 2012, Bart Coninckx wrote: I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? Raspberry Pi would be the obvious choice, surely? It's been done... Someone on the forums is giving away an image that boots into Asterisk - that then connects to their own ITSP server with a small number of free calls - presumably inviting you to pay him money to make more calls ;-) Have to say it's the last thing on my mind to do with my Pi - well, maybe for fun, but not as a comercial on-going system for a company. It's too small (in physiucal size!) and needs a nice box, etc. However it's more than capable although I'd be wary of the speed of the SD card to store voicemail on - it's fast enough, but things like fsync do appear to take a (relatively) long time, making some stuff feel a little clunky on it. Gordon -- _ -- 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] looking for solid state like PC suitable for Asterisk
There seems to even be a 1.6 Ghz Intel Atom device. One site I'm looking to use this for has about 40 SIP phones and three BRIs. It's always a guessing game whether devices like this are up for that. If they do have some processing power, I might even consider combining them as a highly available Asterisk cluster (using DRBD and Pacemaker). Anyone 2 cents about that? BC On 05/10/12 14:28, John Novack wrote: Correct. I have never been accused of being a good speller! JN Bart Coninckx wrote: That's Soekris I suppose. Never heard of them, but it looks mighty interesting. Cheers, BC On 05/10/12 13:35, John Novack wrote: I use HP Thin Clients with AstLinux installed. HP 5720's are available on eBay for not much money, or there are many small boards available new if you don't or can't use used. 10 watts, no fan, no HD Not sure what might be available in your part of the world, but there are Sockris and ALIX flash based boards. AstLinux has special configurations for these. I have 20-30 AstLinux on thin clients working without a belch on a private collectors network John Novack -- _ -- 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] looking for solid state like PC suitable for Asterisk
These prices are simply stunning ... Little can go wrong with the CPU's speed. awesome, BC On 05/10/12 14:32, Terry Brummell wrote: This thread may interest you. Add a SSD and RAM and you're good to go! http://pbxinaflash.com/community/index.php?threads/diy-piaf2-server-200. 12460/ -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Thursday, May 10, 2012 8:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] looking for solid state like PC suitable for Asterisk Correct. I have never been accused of being a good speller! JN Bart Coninckx wrote: That's Soekris I suppose. Never heard of them, but it looks mighty interesting. Cheers, BC On 05/10/12 13:35, John Novack wrote: I use HP Thin Clients with AstLinux installed. HP 5720's are available on eBay for not much money, or there are many small boards available new if you don't or can't use used. 10 watts, no fan, no HD Not sure what might be available in your part of the world, but there are Sockris and ALIX flash based boards. AstLinux has special configurations for these. I have 20-30 AstLinux on thin clients working without a belch on a private collectors network John Novack Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? thx!! BC -- _ -- 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
Re: [asterisk-users] looking for solid state like PC suitable for Asterisk
Tim, looked at these briefly, they all seemed pre-installed, correct? Is reinstallation with, let's say, CentOS possible? thx, BC On 05/10/12 14:39, Tim Nelson wrote: Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? Have a look at the Blackbochs SBC***. It is small, low power, plenty of storage options, built in analog telephony ports, etc: http://www.rockbochs.com/products/blackbochs-sbc --Tim ***Yes, I'm affiliated with the product/company, but it is on topic for this discussion. My apologies if this offends anyone. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Increasing voice volume without getting echo or entered digit problem
Hi Bilal, High volume is always a big for echo cancellation. The problem is that the signal reaches saturation and therefore reduce the effectiveness of the detection/convergence. If your existing echo cancellation can not handle it, you might want to try a different algorithm for echo cancellation. Try the PBXMate to see it resolves the problem in your case. Regards, Valer From: bilal ghayyad bilmar...@yahoo.com To: asterisk-users@lists.digium.com Sent: Thursday, May 10, 2012 2:11 PM Subject: [asterisk-users] Increasing voice volume without getting echo or entered digit problem Dears; How I can increase the voice volume in the analogue line (at dahdi port) without getting a problems in the entered digits (when using Background function)? I got to know previously that it is possible to increase the gain at hardware and not software, this is better to avoid getting a problems. But where? I am using DAHDI 2.4 and another machine has DAHDI 2.6 Using txgain and rxgain from chan_dahdi.conf will not help, it will increase the voice volume but with the following problems: 1) Suddenly the call will be disconnected while we are talking. 2) When calling the Asterisk box and we entered the digits, it is failing to collect it (sometime does not collect it correctly and sometime it collects the digit duplicated). 3) Echo problem. So I need to know how to increase the voice volume from another place? Appreciate the kindly help. Regards Bilal -- _ -- 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] Email-to-Fax
Dear, I am using Fax-to-Email feature of FreePBX, now i am looking Email-To-Fax option with freePBX, kindly update is it possible to have this feature with FreePBX? kindly contact me on my email miana...@msn.com if anybody have this type of solution. thanks. -- Regards, M. Asif Raza -- _ -- 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] looking for solid state like PC suitable for Asterisk
- Original Message - Tim, looked at these briefly, they all seemed pre-installed, correct? Is reinstallation with, let's say, CentOS possible? thx, BC The units *can* come preinstalled with our PBX flavor (Debian, Asterisk, FreePBX), or they can be sent bare and you can install your OS/platform of choice. CentOS specifically does not run on the board as the upstream vendor does not support i586 arch any longer (since Centos 5.x series IIRC). We've done some work trying to patch the installer and use custom kernels to get around this, but were unsuccessful. --Tim -- _ -- 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] SLA - Shared Line Appearance - Polycom
Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom phones. I want to be able to emulate a key system but I cannot figure it out. Everything I tried so far is just not working together. Thanks, --E -- _ -- 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] enabling dialing by sip uri
I have following sip account :
Name/username HostDyn
Forcerport ACL Port Status Description
demo-alice/demo-alice 192.168.7.47 D
N 1080 Unmonitored
demo-bob/demo-bob 192.168.7.47 D
N 5060 Unmonitored
and i have set up the following extensions for them:
ASTERISK_IP=192.168.7.39
[users]
exten=6001,1,Dial(SIP/demo-alice,20)
exten=6002,1,Dial(SIP/demo-bob,20)
exten = _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}]?unhandled)
exten = _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}:5060]?unhandled)
exten = _.,n,Macro(uri-dial,${EXTEN}@${SIPDOMAIN})
exten = _.,n,HangUp()u
[macro-uri-dial]
exten=s,n,NoOp(Calling as SIP address: ${ARG1})
exten=s,n,Dial(SIP/${ARG1},60)
But if i dial sip uri the call does not happen. asterisk cli shows
extension is rejected.
--
-aft
--
_
-- 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] SLA - Shared Line Appearance - Polycom
Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom phones. I want to be able to emulate a key system but I cannot figure it out. Everything I tried so far is just not working together. Thanks, --E -- _ -- 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] SLA - Shared Line Appearance - Polycom
Which flavor of Asterisk and what model Polycom? Both are factors in the possible success of the task. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 9:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SLA - Shared Line Appearance - Polycom Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom phones. I want to be able to emulate a key system but I cannot figure it out. Everything I tried so far is just not working together. Thanks, --E -- _ -- 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] enabling dialing by sip uri
On 05/10/2012 09:39 AM, Arif Hossain wrote:
I have following sip account :
Name/username HostDyn
Forcerport ACL Port Status Description
demo-alice/demo-alice 192.168.7.47 D
N 1080 Unmonitored
demo-bob/demo-bob 192.168.7.47 D
N 5060 Unmonitored
and i have set up the following extensions for them:
ASTERISK_IP=192.168.7.39
[users]
exten=6001,1,Dial(SIP/demo-alice,20)
exten=6002,1,Dial(SIP/demo-bob,20)
exten = _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}]?unhandled)
exten = _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}:5060]?unhandled)
exten = _.,n,Macro(uri-dial,${EXTEN}@${SIPDOMAIN})
exten = _.,n,HangUp()u
[macro-uri-dial]
exten=s,n,NoOp(Calling as SIP address: ${ARG1})
exten=s,n,Dial(SIP/${ARG1},60)
But if i dial sip uri the call does not happen. asterisk cli shows
extension is rejected.
Asterisk is not a SIP proxy. If you are entering a SIP URI into your
phone, and that URI does not resolve to the Asterisk server as its
target, then the INVITE request sent by the phone should not even be
sent to Asterisk at all (it should go to wherever the URI resolves to).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for solid state like PC suitable for Asterisk
On 05/10/2012 03:49 AM, Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Just a small comment here... I really find it quite humorous that people use 'solid state' to mean 'no moving parts'. All of the parts of my computers that move are still composed of solid materials, and the electrical currents involved in them still move through solid materials :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for solid state like PC suitable for Asterisk
- Original Message - On 05/10/2012 03:49 AM, Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Just a small comment here... I really find it quite humorous that people use 'solid state' to mean 'no moving parts'. All of the parts of my computers that move are still composed of solid materials, and the electrical currents involved in them still move through solid materials :-) I think most users are just trying to be specific about not wanting any computer equipment where tubes[1] are in use. :D --Tim (...who still uses and loves his tube audio gear...) [1] http://en.wikipedia.org/wiki/Vacuum_tube -- _ -- 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] looking for solid state like PC suitable for Asterisk
- Original Message - From: Tim Nelson tnel...@rockbochs.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 10, 2012 11:43:07 AM Subject: Re: [asterisk-users] looking for solid state like PC suitable for Asterisk - Original Message - On 05/10/2012 03:49 AM, Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Just a small comment here... I really find it quite humorous that people use 'solid state' to mean 'no moving parts'. All of the parts of my computers that move are still composed of solid materials, and the electrical currents involved in them still move through solid materials :-) I think most users are just trying to be specific about not wanting any computer equipment where tubes[1] are in use. :D --Tim (...who still uses and loves his tube audio gear...) [1] http://en.wikipedia.org/wiki/Vacuum_tube You know... as opposed to liquid or gaseous state. No one wants a PC that will just slide down the drain or disperse into mist. -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SLA - Shared Line Appearance - Polycom
Oh yeah, damn small things. Asterisk 1.8.7.1 Polycom IP650 Thanks, --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 10:47 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Which flavor of Asterisk and what model Polycom? Both are factors in the possible success of the task. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 9:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SLA - Shared Line Appearance - Polycom Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom phones. I want to be able to emulate a key system but I cannot figure it out. Everything I tried so far is just not working together. Thanks, --E -- _ -- 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] looking for solid state like PC suitable for Asterisk
On Thu, 2012-05-10 at 07:40 -0500, Tim Nelson wrote: - Original Message - On Thursday 10 May 2012, Bart Coninckx wrote: I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? Raspberry Pi would be the obvious choice, surely? The hype around the Raspberry Pi is enormous. I would not consider it a real option for production voice until it's had a chance to mature and be available for some time to iron out the bugs, both hardware and software related. My $0.02 USD. Another couple of cents: the pi comes only with arm-cpu and limited amount mem - no upgrade possible. Might be an issue for asterisk... have a look at: http://www.fit-pc.info/ As long as you don't need to plug in a pci-board it is nice small and uses hardly any amps. hw -- _ -- 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] SLA - Shared Line Appearance - Polycom
You read this? - http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/SLA.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 12:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Oh yeah, damn small things. Asterisk 1.8.7.1 Polycom IP650 Thanks, --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 10:47 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Which flavor of Asterisk and what model Polycom? Both are factors in the possible success of the task. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 9:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SLA - Shared Line Appearance - Polycom Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom phones. I want to be able to emulate a key system but I cannot figure it out. Everything I tried so far is just not working together. Thanks, --E -- _ -- 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] SLA - Shared Line Appearance - Polycom
Yes but I am not able to follow it to completion. I think the guide is quite incomplete. I am also trying to follow another guide which leaves things out. I was hoping that someone has it working so they can tell me exactly what is missing and needs to be done. --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 2:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom You read this? - http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/SLA.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 12:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Oh yeah, damn small things. Asterisk 1.8.7.1 Polycom IP650 Thanks, --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 10:47 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Which flavor of Asterisk and what model Polycom? Both are factors in the possible success of the task. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 9:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SLA - Shared Line Appearance - Polycom Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom phones. I want to be able to emulate a key system but I cannot figure it out. Everything I tried so far is just not working together. Thanks, --E -- _ -- 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] SLA - Shared Line Appearance - Polycom
I was able to do it with 1.4 and an IP501. You just get your hints set up correctly and life should be good. You set up the hints as contacts in the directory. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 1:31 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Yes but I am not able to follow it to completion. I think the guide is quite incomplete. I am also trying to follow another guide which leaves things out. I was hoping that someone has it working so they can tell me exactly what is missing and needs to be done. --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 2:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom You read this? - http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/SLA.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 12:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Oh yeah, damn small things. Asterisk 1.8.7.1 Polycom IP650 Thanks, --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 10:47 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Which flavor of Asterisk and what model Polycom? Both are factors in the possible success of the task. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 9:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SLA - Shared Line Appearance - Polycom Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom phones. I want to be able to emulate a key system but I cannot figure it out. Everything I tried so far is just not working together. Thanks, --E -- _ -- 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] Increasing voice volume without getting echo or entered digit problem
Dear; I am talking about something else. When I said increasing the volume from the hardware, I was mean something else. Before when we were using Zaptel, we were able to do this (increasing the volume from the hardware) in the /etc/modprobe.conf but currently we are using dahdi and I do not know from where we can do it. Before, we can do it from modprobe.conf using below command: options wctdm fxorxgain=20.0 fxotxgain=20.0 So how to do this in the dahdi? There is a file /etc/modprobe.d/dahdi.conf, is it the right file? Or there is another file? Please advise. Regards Bilal -- Hi Bilal, High volume is always a big for echo cancellation. The problem is that the signal reaches saturation and therefore reduce the effectiveness of the detection/convergence.? If your existing echo cancellation can not handle it, you might want to try a different algorithm for echo cancellation. Try the PBXMate to see it resolves the problem in your case. Regards, Valer -- _ -- 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] SLA - Shared Line Appearance - Polycom
Unfortunately I am still a sitting duck. Any old configs you can show me? Its driving me insane. I am trying to set it up in this fashion: Everything is SIP based. I do not have ZAP/Dahdi/etc. 6 IP650 phones. line keys 3,4,5,6 will be emulating the Key System with line keys 1 and 2 be set to have their extension. A call comes in, it rings SLA_line 1 ( line key 3) on all phones. If I pick up, that line is now active on all phones. If another call comes in, it goes to SLA_Line 2 ( line key 4 ). If I put SLA_line1 on hold, it shows on hold for all phones. A coworker then picks up SLA_line1 and continues to the conversation from their phone. I do not necessarily care if they can call out on the SLA_line(n)s since they will have line key 1 and 2 set to their personal extension. -E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 2:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom I was able to do it with 1.4 and an IP501. You just get your hints set up correctly and life should be good. You set up the hints as contacts in the directory. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 1:31 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Yes but I am not able to follow it to completion. I think the guide is quite incomplete. I am also trying to follow another guide which leaves things out. I was hoping that someone has it working so they can tell me exactly what is missing and needs to be done. --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 2:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom You read this? - http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/SLA.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 12:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Oh yeah, damn small things. Asterisk 1.8.7.1 Polycom IP650 Thanks, --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 10:47 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Which flavor of Asterisk and what model Polycom? Both are factors in the possible success of the task. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 9:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SLA - Shared Line Appearance - Polycom Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom phones. I want to be able to emulate a key system but I cannot figure it out. Everything I tried so far is just not working together. Thanks, --E -- _ -- 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] Increasing voice volume without getting echo or entered digit problem
/etc/asterisk/chan_dahdi.conf is where you control txgain and rxgain in DAHDI. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Thursday, May 10, 2012 1:37 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Increasing voice volume without getting echo or entered digit problem Dear; I am talking about something else. When I said increasing the volume from the hardware, I was mean something else. Before when we were using Zaptel, we were able to do this (increasing the volume from the hardware) in the /etc/modprobe.conf but currently we are using dahdi and I do not know from where we can do it. Before, we can do it from modprobe.conf using below command: options wctdm fxorxgain=20.0 fxotxgain=20.0 So how to do this in the dahdi? There is a file /etc/modprobe.d/dahdi.conf, is it the right file? Or there is another file? Please advise. Regards Bilal -- Hi Bilal, High volume is always a big for echo cancellation. The problem is that the signal reaches saturation and therefore reduce the effectiveness of the detection/convergence.? If your existing echo cancellation can not handle it, you might want to try a different algorithm for echo cancellation. Try the PBXMate to see it resolves the problem in your case. Regards, Valer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SLA - Shared Line Appearance - Polycom
Since you are doing all SIP, you would want something like this: exten = 1000,hint,SIP/100 exten = 2000,hint,SIP/200 exten = 3000,hint,SIP/300 exten = 4000,hint,SIP/400 Then set up your lines to look for 1000@default, 2000@default, etc. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 1:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Unfortunately I am still a sitting duck. Any old configs you can show me? Its driving me insane. I am trying to set it up in this fashion: Everything is SIP based. I do not have ZAP/Dahdi/etc. 6 IP650 phones. line keys 3,4,5,6 will be emulating the Key System with line keys 1 and 2 be set to have their extension. A call comes in, it rings SLA_line 1 ( line key 3) on all phones. If I pick up, that line is now active on all phones. If another call comes in, it goes to SLA_Line 2 ( line key 4 ). If I put SLA_line1 on hold, it shows on hold for all phones. A coworker then picks up SLA_line1 and continues to the conversation from their phone. I do not necessarily care if they can call out on the SLA_line(n)s since they will have line key 1 and 2 set to their personal extension. -E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 2:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom I was able to do it with 1.4 and an IP501. You just get your hints set up correctly and life should be good. You set up the hints as contacts in the directory. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 1:31 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Yes but I am not able to follow it to completion. I think the guide is quite incomplete. I am also trying to follow another guide which leaves things out. I was hoping that someone has it working so they can tell me exactly what is missing and needs to be done. --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 2:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom You read this? - http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/SLA.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 12:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Oh yeah, damn small things. Asterisk 1.8.7.1 Polycom IP650 Thanks, --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 10:47 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Which flavor of Asterisk and what model Polycom? Both are factors in the possible success of the task. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 9:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SLA - Shared Line Appearance - Polycom Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom phones. I want to be able to emulate a key system but I cannot figure it out. Everything I tried so far is just not working together. Thanks, --E -- _ -- 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] Increasing voice volume without getting echo or entered digit problem
On Thu, May 10, 2012 at 11:36:51AM -0700, bilal ghayyad wrote: When I said increasing the volume from the hardware, I was mean something else. Before when we were using Zaptel, we were able to do this (increasing the volume from the hardware) in the /etc/modprobe.conf but currently we are using dahdi and I do not know from where we can do it. Before, we can do it from modprobe.conf using below command: options wctdm fxorxgain=20.0 fxotxgain=20.0 So how to do this in the dahdi? There is a file /etc/modprobe.d/dahdi.conf, is it the right file? Or there is another file? Yes, that is the correct file. Any *.conf file in /etc/modprobe.d will suffice but dahdi.conf is the convention. You can also accomplish this from the Asterisk CLI with dahdi set hwgain. Type dahdi set hwgain on the Asterisk CLI for more information. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SLA - Shared Line Appearance - Polycom
Would those be the shared lines? Continuing with the assumption they are, I need to set them up in stations in the SLA.conf file? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 2:44 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Since you are doing all SIP, you would want something like this: exten = 1000,hint,SIP/100 exten = 2000,hint,SIP/200 exten = 3000,hint,SIP/300 exten = 4000,hint,SIP/400 Then set up your lines to look for 1000@default, 2000@default, etc. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 1:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Unfortunately I am still a sitting duck. Any old configs you can show me? Its driving me insane. I am trying to set it up in this fashion: Everything is SIP based. I do not have ZAP/Dahdi/etc. 6 IP650 phones. line keys 3,4,5,6 will be emulating the Key System with line keys 1 and 2 be set to have their extension. A call comes in, it rings SLA_line 1 ( line key 3) on all phones. If I pick up, that line is now active on all phones. If another call comes in, it goes to SLA_Line 2 ( line key 4 ). If I put SLA_line1 on hold, it shows on hold for all phones. A coworker then picks up SLA_line1 and continues to the conversation from their phone. I do not necessarily care if they can call out on the SLA_line(n)s since they will have line key 1 and 2 set to their personal extension. -E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 2:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom I was able to do it with 1.4 and an IP501. You just get your hints set up correctly and life should be good. You set up the hints as contacts in the directory. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 1:31 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Yes but I am not able to follow it to completion. I think the guide is quite incomplete. I am also trying to follow another guide which leaves things out. I was hoping that someone has it working so they can tell me exactly what is missing and needs to be done. --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 2:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom You read this? - http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/SLA.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 12:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Oh yeah, damn small things. Asterisk 1.8.7.1 Polycom IP650 Thanks, --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 10:47 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Which flavor of Asterisk and what model Polycom? Both are factors in the possible success of the task. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 9:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SLA - Shared Line Appearance - Polycom Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom phones. I want to be able to emulate a key system but I cannot figure it out. Everything I tried so far is just not working together. Thanks, --E -- _ -- 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] SLA - Shared Line Appearance - Polycom
Correct - the sla.conf makes the line active the hint makes it accessible by the phone. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 1:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Would those be the shared lines? Continuing with the assumption they are, I need to set them up in stations in the SLA.conf file? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 2:44 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Since you are doing all SIP, you would want something like this: exten = 1000,hint,SIP/100 exten = 2000,hint,SIP/200 exten = 3000,hint,SIP/300 exten = 4000,hint,SIP/400 Then set up your lines to look for 1000@default, 2000@default, etc. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 1:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Unfortunately I am still a sitting duck. Any old configs you can show me? Its driving me insane. I am trying to set it up in this fashion: Everything is SIP based. I do not have ZAP/Dahdi/etc. 6 IP650 phones. line keys 3,4,5,6 will be emulating the Key System with line keys 1 and 2 be set to have their extension. A call comes in, it rings SLA_line 1 ( line key 3) on all phones. If I pick up, that line is now active on all phones. If another call comes in, it goes to SLA_Line 2 ( line key 4 ). If I put SLA_line1 on hold, it shows on hold for all phones. A coworker then picks up SLA_line1 and continues to the conversation from their phone. I do not necessarily care if they can call out on the SLA_line(n)s since they will have line key 1 and 2 set to their personal extension. -E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 2:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom I was able to do it with 1.4 and an IP501. You just get your hints set up correctly and life should be good. You set up the hints as contacts in the directory. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 1:31 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Yes but I am not able to follow it to completion. I think the guide is quite incomplete. I am also trying to follow another guide which leaves things out. I was hoping that someone has it working so they can tell me exactly what is missing and needs to be done. --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 2:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom You read this? - http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/SLA.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 12:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Oh yeah, damn small things. Asterisk 1.8.7.1 Polycom IP650 Thanks, --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 10:47 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Which flavor of Asterisk and what model Polycom? Both are factors in the possible success of the task. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 9:45 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SLA - Shared Line Appearance - Polycom Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom phones. I want to be able to emulate a key system but I cannot figure it out. Everything I tried so far is just not working together. Thanks, --E -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --
Re: [asterisk-users] SLA - Shared Line Appearance - Polycom
Then on my polycom what are the configurations? Is it with the line keys such as reg.1 be my extension with line key set to 2 and reg.2 be 1000 ( SLA line 1 ) reg.3 = 200 ( SLA line 2 ) Or is this set up in the bottom part of the configs under the attendant section? --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 2:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Correct - the sla.conf makes the line active the hint makes it accessible by the phone. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 1:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Would those be the shared lines? Continuing with the assumption they are, I need to set them up in stations in the SLA.conf file? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 2:44 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Since you are doing all SIP, you would want something like this: exten = 1000,hint,SIP/100 exten = 2000,hint,SIP/200 exten = 3000,hint,SIP/300 exten = 4000,hint,SIP/400 Then set up your lines to look for 1000@default, 2000@default, etc. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 1:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Unfortunately I am still a sitting duck. Any old configs you can show me? Its driving me insane. I am trying to set it up in this fashion: Everything is SIP based. I do not have ZAP/Dahdi/etc. 6 IP650 phones. line keys 3,4,5,6 will be emulating the Key System with line keys 1 and 2 be set to have their extension. A call comes in, it rings SLA_line 1 ( line key 3) on all phones. If I pick up, that line is now active on all phones. If another call comes in, it goes to SLA_Line 2 ( line key 4 ). If I put SLA_line1 on hold, it shows on hold for all phones. A coworker then picks up SLA_line1 and continues to the conversation from their phone. I do not necessarily care if they can call out on the SLA_line(n)s since they will have line key 1 and 2 set to their personal extension. -E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 2:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom I was able to do it with 1.4 and an IP501. You just get your hints set up correctly and life should be good. You set up the hints as contacts in the directory. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 1:31 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Yes but I am not able to follow it to completion. I think the guide is quite incomplete. I am also trying to follow another guide which leaves things out. I was hoping that someone has it working so they can tell me exactly what is missing and needs to be done. --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 2:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom You read this? - http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/SLA.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr Sent: Thursday, May 10, 2012 12:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Oh yeah, damn small things. Asterisk 1.8.7.1 Polycom IP650 Thanks, --E From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, May 10, 2012 10:47 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] SLA - Shared Line Appearance - Polycom Which flavor of Asterisk and what model Polycom? Both are factors in the possible success of the task. From:
Re: [asterisk-users] British Telecom ISDN BRI line issues
Hi All, I did downgrade to asterisk 1.6.2.6 and dahdi 2.3 which is working with other BRI line (in our NL office), but I get this type of errores: -- Called G1/0788744550 [May 10 20:07:20] ERROR[27479]: chan_dahdi.c:12280 dahdi_pri_error: ACK received for '1' outside of window of '0' to '0', restarting == Primary D-Channel on span 3 down [May 10 20:07:20] WARNING[27479]: chan_dahdi.c:4158 pri_find_dchan: No D-channels available! Using Primary channel 9 as D-channel anyway! == Primary D-Channel on span 3 up [May 10 20:07:20] ERROR[27480]: chan_dahdi.c:12280 dahdi_pri_error: Huh!? no master found I didn't change my config in my previous post, anyone familiar with this type of errors? On Wed, May 9, 2012 at 3:09 PM, khalid touati khalidtou...@gmail.comwrote: Yeah they have a wonderful policy that says ISDN team are not contactable :( thanks a lot!! On Wed, May 9, 2012 at 3:06 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 09-05-12 20:57, khalid touati wrote: Yeah sorry for that, I realized something is missing after I sent the email, but it is exactly what I have (other than order here, which doesn't really matter: you posted ami,te,term, I have ami,term,te). Actually I had couple technicians from digium look at it and they said BT equipements is not responding to the card within a certain range that the card is looking for (i'm not sure what range but I do believe too it's a BT issue), But I have run all the couple command that Patrick suggested (to double check), tested again and still same kind of errors. But Thank you very much Patrick for the guide, I was looking for that it's been a couple days!! I just hope someone that has the exact same issue or someone with previous BT experience see this and help :) ..we never know :) ! Too bad you could not (yet) make it work. Hope you get somewhere with BT. Once you get past the people following those silly scripts you should be able to talk to someone who has a clue and resolve this issue. Good luck! Regards, Patrick -- _ -- 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 -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- _ -- 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] Increasing voice volume without getting echo or entered digit problem
I am sure there should be another place .. if I increased it from chan_dahdi.conf, the voice quality is bad and the calls will disconnecting while we are talking .. Increasing voice volume from chan_dahdi means increasing it at software level, I am sure there is a place to increase it at hardware level. Let us agree on something: Is settings to increase it at hardware level? In Zapata, it was existed and can be done as mentioned in my previous emails (from modprobe.conf), can we agree on this? If yes, so why it is not possible in dahdi? Regards Bilal /etc/asterisk/chan_dahdi.conf is where you control txgain and rxgain in DAHDI. -Original Message- Dear; I am talking about something else. When I said increasing the volume from the hardware, I was mean something else. Before when we were using Zaptel, we were able to do this (increasing the volume from the hardware) in the /etc/modprobe.conf but currently we are using dahdi and I do not know from where we can do it. Before, we can do it from modprobe.conf using below command: options wctdm fxorxgain=20.0 fxotxgain=20.0 So how to do this in the dahdi? There is a file /etc/modprobe.d/dahdi.conf, is it the right file? Or there is another file? Please advise. Regards Bilal -- Hi Bilal, High volume is always a big for echo cancellation. The problem is that the signal reaches saturation and therefore reduce the effectiveness of the detection/convergence.? If your existing echo cancellation can not handle it, you might want to try a different algorithm for echo cancellation. Try the PBXMate to see it resolves the problem in your case. Regards, Valer -- _ -- 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] British Telecom ISDN BRI line issues
On 10-05-12 21:10, khalid touati wrote: Hi All, I did downgrade to asterisk 1.6.2.6 and dahdi 2.3 which is working with other BRI line (in our NL office), but I get this type of errores: -- Called G1/0788744550 [May 10 20:07:20] ERROR[27479]: chan_dahdi.c:12280 dahdi_pri_error: ACK received for '1' outside of window of '0' to '0', restarting == Primary D-Channel on span 3 down [May 10 20:07:20] WARNING[27479]: chan_dahdi.c:4158 pri_find_dchan: No D-channels available! Using Primary channel 9 as D-channel anyway! == Primary D-Channel on span 3 up [May 10 20:07:20] ERROR[27480]: chan_dahdi.c:12280 dahdi_pri_error: Huh!? no master found I didn't change my config in my previous post, anyone familiar with this type of errors? No but there is a bug report with a lot of information that seems similar: https://issues.asterisk.org/jira/browse/14031 In Europe telco's drop the D-channel (cut off power) to save on the electric bill. The libpri/dahdi/asterisk combo should detect a dropped D-channel and signal the telco to fire up the D-channel. Judging from that bugreport (Unresolved) it seems Digium has still not succeeded in properly handling this situation. Should you not be able to resolve this issue and really require an ISDN BRI connection then have a look at an Eicon Diva or Sangoma card. Both cards+drivers properly handle a dropped D-channel. I have used Eicon Diva and Sangoma BRI cards in the Netherlands and they work fine. Or you could get a blackbox from Beronet, Patton, Audiocodes etc. If you feel adventurous you can also get a BRI card with a HFC-S Cologne chipset and get the latest and greatest isdn4k-utils, mISDN, mISDNuser and lcr from http://misdn.eu, build, install and configure lcr to talk to asterisk. A few weeks ago I set it up and did one test call and that call worked fine. Use at own risk :) Regards, Patrick -- _ -- 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] British Telecom ISDN BRI line issues
Thank you Patrick for the detailed info, it does make perfect sense to me, I never expected that Digium cards have such an problem! On Thu, May 10, 2012 at 4:13 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 10-05-12 21:10, khalid touati wrote: Hi All, I did downgrade to asterisk 1.6.2.6 and dahdi 2.3 which is working with other BRI line (in our NL office), but I get this type of errores: -- Called G1/0788744550 [May 10 20:07:20] ERROR[27479]: chan_dahdi.c:12280 dahdi_pri_error: ACK received for '1' outside of window of '0' to '0', restarting == Primary D-Channel on span 3 down [May 10 20:07:20] WARNING[27479]: chan_dahdi.c:4158 pri_find_dchan: No D-channels available! Using Primary channel 9 as D-channel anyway! == Primary D-Channel on span 3 up [May 10 20:07:20] ERROR[27480]: chan_dahdi.c:12280 dahdi_pri_error: Huh!? no master found I didn't change my config in my previous post, anyone familiar with this type of errors? No but there is a bug report with a lot of information that seems similar: https://issues.asterisk.org/jira/browse/14031 In Europe telco's drop the D-channel (cut off power) to save on the electric bill. The libpri/dahdi/asterisk combo should detect a dropped D-channel and signal the telco to fire up the D-channel. Judging from that bugreport (Unresolved) it seems Digium has still not succeeded in properly handling this situation. Should you not be able to resolve this issue and really require an ISDN BRI connection then have a look at an Eicon Diva or Sangoma card. Both cards+drivers properly handle a dropped D-channel. I have used Eicon Diva and Sangoma BRI cards in the Netherlands and they work fine. Or you could get a blackbox from Beronet, Patton, Audiocodes etc. If you feel adventurous you can also get a BRI card with a HFC-S Cologne chipset and get the latest and greatest isdn4k-utils, mISDN, mISDNuser and lcr from http://misdn.eu, build, install and configure lcr to talk to asterisk. A few weeks ago I set it up and did one test call and that call worked fine. Use at own risk :) Regards, Patrick -- _ -- 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 -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- _ -- 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] enabling dialing by sip uri
On Thu, May 10, 2012 at 11:50 AM, Kevin P. Fleming kpflem...@digium.com wrote:
On 05/10/2012 09:39 AM, Arif Hossain wrote:
I have following sip account :
Name/username Host Dyn
Forcerport ACL Port Status Description
demo-alice/demo-alice 192.168.7.47 D
N 1080 Unmonitored
demo-bob/demo-bob 192.168.7.47 D
N 5060 Unmonitored
and i have set up the following extensions for them:
ASTERISK_IP=192.168.7.39
[users]
exten=6001,1,Dial(SIP/demo-alice,20)
exten=6002,1,Dial(SIP/demo-bob,20)
exten = _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}]?unhandled)
exten = _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}:5060]?unhandled)
exten = _.,n,Macro(uri-dial,${EXTEN}@${SIPDOMAIN})
exten = _.,n,HangUp()u
[macro-uri-dial]
exten=s,n,NoOp(Calling as SIP address: ${ARG1})
exten=s,n,Dial(SIP/${ARG1},60)
But if i dial sip uri the call does not happen. asterisk cli shows
extension is rejected.
Asterisk is not a SIP proxy. If you are entering a SIP URI into your phone,
and that URI does not resolve to the Asterisk server as its target, then the
INVITE request sent by the phone should not even be sent to Asterisk at all
(it should go to wherever the URI resolves to).
I'm using the asterisk's ip to form sip uri at the sip client. So it
resolves to asterisk no doubt.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
-aft
--
_
-- 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] British Telecom ISDN BRI line issues
On 05/10/2012 03:20 PM, khalid touati wrote: Thank you Patrick for the detailed info, it does make perfect sense to me, I never expected that Digium cards have such an problem! There are patches in the works already (being tested by users in Europe) to deal with this layer 1 issue. Upcoming releases of DAHDI and Asterisk should have support for it. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] enabling dialing by sip uri
On 05/10/2012 03:36 PM, Arif Hossain wrote: Asterisk is not a SIP proxy. If you are entering a SIP URI into your phone, and that URI does not resolve to the Asterisk server as its target, then the INVITE request sent by the phone should not even be sent to Asterisk at all (it should go to wherever the URI resolves to). I'm using the asterisk's ip to form sip uri at the sip client. So it resolves to asterisk no doubt. You'll have to provide more details (primarily a CLI log) then in order for anyone to be able to help you. You said that Asterisk shows extension is rejected, but extensions don't get rejected. Extensions can be 'not found', but that's very different from rejected. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for solid state like PC suitable for Asterisk
On 05/10/12 18:38, Kevin P. Fleming wrote: On 05/10/2012 03:49 AM, Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Just a small comment here... I really find it quite humorous that people use 'solid state' to mean 'no moving parts'. All of the parts of my computers that move are still composed of solid materials, and the electrical currents involved in them still move through solid materials :-) Yeah, well, have you seen crawling any bugs in software lately? Still they are called bugs ... :-s -- _ -- 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] looking for solid state like PC suitable for Asterisk
- Original Message - On 05/10/12 18:38, Kevin P. Fleming wrote: On 05/10/2012 03:49 AM, Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Just a small comment here... I really find it quite humorous that people use 'solid state' to mean 'no moving parts'. All of the parts of my computers that move are still composed of solid materials, and the electrical currents involved in them still move through solid materials :-) Yeah, well, have you seen crawling any bugs in software lately? Still they are called bugs ... :-s Funny, I've heard them referred to as 'features'. :D --Tim -- _ -- 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] enabling dialing by sip uri
On Thu, May 10, 2012 at 5:52 PM, Kevin P. Fleming kpflem...@digium.com wrote: You'll have to provide more details (primarily a CLI log) then in order for anyone to be able to help you. You said that Asterisk shows extension is rejected, but extensions don't get rejected. Extensions can be 'not found', but that's very different from rejected. Ok i will post more detailed log. -- -aft -- _ -- 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] Event response (AMI)
When I execute the Action commands set then the Event would response back. How would I know which Action are they belong/reference to? For example: ACTION: Originate Channel: SIP/test Exten: 215 Timeout: 3 Context: test Priority: 1 ActionID: 1333 Response: Success ActionID: 1333 Message: Originate successfully queued Event response when I hang up the call: Event: Hangup Privilege: call,all Channel: SIP/test-007f Uniqueid: 1336690030.189 CallerIDNum: unknown CallerIDName: unknown Cause: 16 Cause-txt: Normal Clearing As you can see, how would I know which which ACTION was that belong to? If I were coding in PHP (AMI) to Originate the calls then I want to detect which call hanged up. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for solid state like PC suitable for Asterisk
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Thursday, May 10, 2012 2:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] looking for solid state like PC suitable for Asterisk - Original Message - On 05/10/12 18:38, Kevin P. Fleming wrote: On 05/10/2012 03:49 AM, Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Just a small comment here... I really find it quite humorous that people use 'solid state' to mean 'no moving parts'. All of the parts of my computers that move are still composed of solid materials, and the electrical currents involved in them still move through solid materials :-) Yeah, well, have you seen crawling any bugs in software lately? Still they are called bugs ... :-s Funny, I've heard them referred to as 'features'. :D --Tim -- that's 'undocumented features'... :) -Ric -- _ -- 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] British Telecom ISDN BRI line issues
On 10-05-12 23:47, Kevin P. Fleming wrote: On 05/10/2012 03:20 PM, khalid touati wrote: Thank you Patrick for the detailed info, it does make perfect sense to me, I never expected that Digium cards have such an problem! There are patches in the works already (being tested by users in Europe) to deal with this layer 1 issue. Upcoming releases of DAHDI and Asterisk should have support for it. Thanks for the update Kevin. That's good to know. I look forward to the new releases. Regards, Patrick -- _ -- 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] British Telecom ISDN BRI line issues
Thank you Kevin! thanks Patrickhope a new release will come out soon! On Thu, May 10, 2012 at 7:37 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 10-05-12 23:47, Kevin P. Fleming wrote: On 05/10/2012 03:20 PM, khalid touati wrote: Thank you Patrick for the detailed info, it does make perfect sense to me, I never expected that Digium cards have such an problem! There are patches in the works already (being tested by users in Europe) to deal with this layer 1 issue. Upcoming releases of DAHDI and Asterisk should have support for it. Thanks for the update Kevin. That's good to know. I look forward to the new releases. Regards, Patrick -- _ -- 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 -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- _ -- 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] British Telecom ISDN BRI line issues
Patrick, I got confused though is this true: Any Asterisk soft+digium hdw = it doesn't work Any Asterisk soft+sangoma hdw = it works Patched asterisk soft+digium hdw = it will work (per Kevin) On May 10, 2012 9:06 PM, khalid touati khalidtou...@gmail.com wrote: Thank you Kevin! thanks Patrickhope a new release will come out soon! On Thu, May 10, 2012 at 7:37 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 10-05-12 23:47, Kevin P. Fleming wrote: On 05/10/2012 03:20 PM, khalid touati wrote: Thank you Patrick for the detailed info, it does make perfect sense to me, I never expected that Digium cards have such an problem! There are patches in the works already (being tested by users in Europe) to deal with this layer 1 issue. Upcoming releases of DAHDI and Asterisk should have support for it. Thanks for the update Kevin. That's good to know. I look forward to the new releases. Regards, Patrick -- _ -- 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 -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- _ -- 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] British Telecom ISDN BRI line issues
Hi Khalid, Judging from that bug report I *think*: On 11-05-12 03:39, khalid touati wrote: Patrick, I got confused though is this true: Any Asterisk soft+digium hdw = it doesn't work There seem to be combinations that do work. It is my understanding from that bugreport that an older libpri works with an older version of asterisk that does not have this issue. If your goal is to deploy the latest-and-greatest libpri, dahdi and asterisk 1.8 releases then it does not seem to work. Any Asterisk soft+sangoma hdw = it works In my experience yes. Same goes for Eicon Diva Server cards. Patched asterisk soft+digium hdw = it will work (per Kevin) Yes per Kevin's comment. Regards, Patrick -- _ -- 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] Can run from shell but not from Asterisk System command
Hi All, I have this strange problem on a newly installed PBX. 1.8.12.0. I have other installs of 1.8.12.0 that does not exhibit this problem. I can run from the console /usr/sbin/fax2mail --cid-name Console Execuation --cid-number 61123123 --dest-exten 8960 -f /var/spool/asterisk/fax/1336702728.59 ... and an email will be emailed to me. The following does not produce an email. exten = 1122,1,Answer exten = 1122,n,System(/usr/sbin/fax2mail --cid-name Console Execuation --cid-number 61123123 --dest-exten 8960 -f /var/spool/asterisk/fax/1336702728.59) exten = 1122,n,HangUp result from CLI == Using SIP RTP CoS mark 5 -- Executing [1122@internal:1] Answer(SIP/8930-002b, ) in new stack -- Executing [1122@internal:2] System(SIP/8930-002b, /usr/sbin/fax2mail --cid-name Console Execuation --cid-number 61735108900 --dest-exten 8960 -f /var/spool/asterisk/fax/1336702728.59) in new stack -- Executing [1122@internal:3] Hangup(SIP/8930-002b, ) in new stack == Spawn extension (internal, 1122, 3) exited non-zero on 'SIP/8930-002b' The log files that the fax2mail script generates are all correct but no email. Regards David Klaverstyn -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can run from shell but not from Asterisk System command
Sorry I meant to mention that Asterisk is running as root. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaverstyn, David C Sent: Friday, 11 May 2012 3:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Can run from shell but not from Asterisk System command Hi All, I have this strange problem on a newly installed PBX. 1.8.12.0. I have other installs of 1.8.12.0 that does not exhibit this problem. I can run from the console /usr/sbin/fax2mail --cid-name Console Execuation --cid-number 61123123 --dest-exten 8960 -f /var/spool/asterisk/fax/1336702728.59 ... and an email will be emailed to me. The following does not produce an email. exten = 1122,1,Answer exten = 1122,n,System(/usr/sbin/fax2mail --cid-name Console Execuation --cid-number 61123123 --dest-exten 8960 -f /var/spool/asterisk/fax/1336702728.59) exten = 1122,n,HangUp result from CLI == Using SIP RTP CoS mark 5 -- Executing [1122@internal:1] Answer(SIP/8930-002b, ) in new stack -- Executing [1122@internal:2] System(SIP/8930-002b, /usr/sbin/fax2mail --cid-name Console Execuation --cid-number 61735108900 --dest-exten 8960 -f /var/spool/asterisk/fax/1336702728.59) in new stack -- Executing [1122@internal:3] Hangup(SIP/8930-002b, ) in new stack == Spawn extension (internal, 1122, 3) exited non-zero on 'SIP/8930-002b' The log files that the fax2mail script generates are all correct but no email. Regards David Klaverstyn -- _ -- 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] British Telecom ISDN BRI line issues
Thank you for your reply Patrick! for the first situation, I did try asterisk 1.6.2.6 and dahdi 2.3 but with no success. Can anyone suggest a combination that works till a patch is released? On Thu, May 10, 2012 at 10:48 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: Hi Khalid, Judging from that bug report I *think*: On 11-05-12 03:39, khalid touati wrote: Patrick, I got confused though is this true: Any Asterisk soft+digium hdw = it doesn't work There seem to be combinations that do work. It is my understanding from that bugreport that an older libpri works with an older version of asterisk that does not have this issue. If your goal is to deploy the latest-and-greatest libpri, dahdi and asterisk 1.8 releases then it does not seem to work. Any Asterisk soft+sangoma hdw = it works In my experience yes. Same goes for Eicon Diva Server cards. Patched asterisk soft+digium hdw = it will work (per Kevin) Yes per Kevin's comment. Regards, Patrick -- _ -- 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 -- Khalid Touati Network Administrator at Endosoft, LLC CCNA -- _ -- 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] looking for solid state like PC suitable for Asterisk
2012/5/10, A J Stiles asterisk_l...@earthshod.co.uk: On Thursday 10 May 2012, Bart Coninckx wrote: I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? Raspberry Pi would be the obvious choice, surely? What about G729 and Raspberry Pi ? -- AJS Answers come *after* questions. -- _ -- 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
