[asterisk-users] Faxing success rate on PRI
Hi List, I've been using PSTN-ATA + Asterisk + IAXModem + Hylafax since three years on my lab test setup and I appreciate it. Moreover the global quantity of fax handled by this setup is not very high. I'll be involved in a more complex system for a customer and I would like to ask to All of you if you have experiences and/or statistical results on faxing success and failure rate. The system I have to deploy will operate in the following context: - It will be interfaced to an E1 PRI - It will be able to send and receive faxes (by e-mail and/or virtual printers) - It will be able to send faxes from a normal fax machine. The system will be placed on the same building, i.e. only private ethernet trunks. I'm thinking to this type of solution: - Patton external unit for E1 - Asterisk 1.4 + IAXModem + Hylafax - An external ATA for the fax machine but I'm open to any other possible solution (I'm thinking to have a demodulation on Patton and talk T38 with Asterisk 1.6). The fax volume will be high because actually the customer has a ZFax software system with 12 fax-modem installed (that will be replaced by the system). I know that this was already asked in this list in the past, but I would like to know if someone has experience on this and could share their opinion, tricks and/or statistical results on failure/success rate when faxing. I think that this could be useful to other people have to realize a system like that one depicted. Thank you in advance. Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Communicator with Asterisk/Trixbox
Is it the Windows software, or other? I noticed the Nokia E71 mobile has an option for Cisco IP Communicator (besides the built-in SIP client) On Wed, Mar 4, 2009 at 22:32, Dorien K. Takeshi dorien.take...@webhad.co.nz wrote: Hi guys, Has anyone had any luck with getting the Cisco IP Communicator working with your Asterisk or primarily, Trixbox installation? I've tried searching the net for information, and found someone said to set it up like the 7970 hard phone, which I have, and I'm just running into the problems with it saying Error Verifying Config Info. Any and all help is appreciated. Dorien ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing success rate on PRI
Here is my current setup: E1 = [Asterisk with TE220p] = IAX Trunk (routed network) = [Asterisk with TDM800p] = Fax/Copy Machine This seems to work fine, a few failed Fax or very slow sending/process sometimes but no complaining users, so this must be ok :) My previous try was: E1 = [Asterisk with TE220p] = IAX peer (same network) = HylaFax machine with IAXmodem While the users did like the facility of receiving PDF documents, we had many calls for people unable to send Fax to us, so i kinda ditched to solution. I've updated the setup to use IAXmodem/Hylafax on the first asterisk machine but until now i haven't tested it. Marco a écrit : Hi List, I've been using PSTN-ATA + Asterisk + IAXModem + Hylafax since three years on my lab test setup and I appreciate it. Moreover the global quantity of fax handled by this setup is not very high. I'll be involved in a more complex system for a customer and I would like to ask to All of you if you have experiences and/or statistical results on faxing success and failure rate. The system I have to deploy will operate in the following context: - It will be interfaced to an E1 PRI - It will be able to send and receive faxes (by e-mail and/or virtual printers) - It will be able to send faxes from a normal fax machine. The system will be placed on the same building, i.e. only private ethernet trunks. I'm thinking to this type of solution: - Patton external unit for E1 - Asterisk 1.4 + IAXModem + Hylafax - An external ATA for the fax machine but I'm open to any other possible solution (I'm thinking to have a demodulation on Patton and talk T38 with Asterisk 1.6). The fax volume will be high because actually the customer has a ZFax software system with 12 fax-modem installed (that will be replaced by the system). I know that this was already asked in this list in the past, but I would like to know if someone has experience on this and could share their opinion, tricks and/or statistical results on failure/success rate when faxing. I think that this could be useful to other people have to realize a system like that one depicted. Thank you in advance. Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing success rate on PRI
On Sun, 8 Mar 2009, Marco wrote: Hi List, I've been using PSTN-ATA + Asterisk + IAXModem + Hylafax since three years on my lab test setup and I appreciate it. Moreover the global quantity of fax handled by this setup is not very high. I'll be involved in a more complex system for a customer and I would like to ask to All of you if you have experiences and/or statistical results on faxing success and failure rate. The system I have to deploy will operate in the following context: - It will be interfaced to an E1 PRI - It will be able to send and receive faxes (by e-mail and/or virtual printers) - It will be able to send faxes from a normal fax machine. The system will be placed on the same building, i.e. only private ethernet trunks. I'm thinking to this type of solution: - Patton external unit for E1 Out of curiosity, why an external box rather than something like a TE120P PCI card? - Asterisk 1.4 + IAXModem + Hylafax - An external ATA for the fax machine but I'm open to any other possible solution (I'm thinking to have a demodulation on Patton and talk T38 with Asterisk 1.6). Personally, I think you're adding complexity and can't see why that would be better than an on-board PRI card... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing success rate on PRI
On Sun, 8 Mar 2009, benoit wrote: Here is my current setup: E1 = [Asterisk with TE220p] = IAX Trunk (routed network) = [Asterisk with TDM800p] = Fax/Copy Machine The TE220P and the TDM800P are in different Asterisk boxes? Any particular reason for that? I now have an E1 coming in to the asterisk box and IAXmodem and HylaFAX to receive faxes which works flawlessly. Outbound i still have an old faxserver connected to an old analogue line though. My faxing needs require that i have a hardcopy printout of every fax that is sent like a reduced size of the fax sent on one A4 together with all status info (Fax number it was sent to, status, duration, ID of the receiver, date, time etc.). So far i couldn't find any solution that does that. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing success rate on PRI
Remco Barendse a écrit : On Sun, 8 Mar 2009, benoit wrote: Here is my current setup: E1 = [Asterisk with TE220p] = IAX Trunk (routed network) = [Asterisk with TDM800p] = Fax/Copy Machine The TE220P and the TDM800P are in different Asterisk boxes? Any particular reason for that? Well the Fax machine and the E1 input are wired in two differents locations, and at the time of the setup all links between the two servers rooms where used for networks and other stuff. Also the first asterisk already had a TE220 and a B410p and i feared adding one more card wouldn't be a great idea. I now have an E1 coming in to the asterisk box and IAXmodem and HylaFAX to receive faxes which works flawlessly. Outbound i still have an old faxserver connected to an old analogue line though. My faxing needs require that i have a hardcopy printout of every fax that is sent like a reduced size of the fax sent on one A4 together with all status info (Fax number it was sent to, status, duration, ID of the receiver, date, time etc.). So far i couldn't find any solution that does that. Couldn't you hack a faxsend script for hylafax to do this ? my first setup involved sending all fax by mail to some people in pdf form but to speed up filtering spam all, i also sent the first page of the inbound fax in a jpg image so it appear directly in the e-mail ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compile problems
In makemenuconfig at apps you have rxfax and txfax you can disable them from there . I think the name is app-fax i dont remember exactly now but it is there Enviado desde mi iPhone El 08/03/2009, a las 04:33 a.m., Remco Barendse aster...@barendse.to escribió: On Sun, 8 Mar 2009, Sebastian wrote: The fax error seems to be problem of spandsp version. What version are you using??? I use the latest IAXMODEM 1.2.0, the changelog of it says update spandsp to 20080725 snapshot However, i never asked Asterisk to compile with fax support, can i disable fax support somewhere? make menuconfig for asterisk didn't get me much further ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Server Setup Advice
Hello Everybody! I am currently setting up an Asterisk server for medium to high load (approximately 20-35 concurrent phone lines). Do you think the following specs will sufficiently satisfy this system? CPU: XeonQC3220 2.4GHZ 8M RAM: 2X2GB/800 Harddrive: 1X250GB I could add harddrives and partition them into /var and /log directories to help with diskdrive throughput. Thanks! Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing success rate on PRI
benoit wrote: Remco Barendse a écrit : On Sun, 8 Mar 2009, benoit wrote: Here is my current setup: E1 = [Asterisk with TE220p] = IAX Trunk (routed network) = [Asterisk with TDM800p] = Fax/Copy Machine You'll find that faxing over IAX is problematic at best. If this is your only option, I'd suggest that you have HylaFAX+ running at both ends. Capture the faxes coming off of the E1 as PDF, do a remote send to the other fax server and have the remote HylaFAX+ do the faxing to the fax/copy machine. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Simple Meetme Question
Hello, setting up Meetme was very easy. I jut added the MeetMe Application to an internal extension to be reachable by SIP and to an external extension to be reachable by ISDN. What I don't understand however is how to call somebody and drop him to the conference? I'm using Asterisk 1.4 from Debian lenny Sven -- In the land of the brave and the free, we defend our freedom with the GNU GPL (Richard M. Stallman on www.gnu.org) /me is gig...@ircnet, http://sven.gegg.us/ on the Web ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing success rate on PRI
Doug Lytle a écrit : benoit wrote: Remco Barendse a écrit : On Sun, 8 Mar 2009, benoit wrote: Here is my current setup: E1 = [Asterisk with TE220p] = IAX Trunk (routed network) = [Asterisk with TDM800p] = Fax/Copy Machine You'll find that faxing over IAX is problematic at best. If this is your only option, I'd suggest that you have HylaFAX+ running at both ends. Capture the faxes coming off of the E1 as PDF, do a remote send to the other fax server and have the remote HylaFAX+ do the faxing to the fax/copy machine. Doug Well, as i said before this setup works quite fine now. What doesn't was when using hylafax over IAXmodem on two separate servers. And hardware faxes make a much better job at handling faulty communication than hylafax ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Setup Advice
Look like good. I have an similar server for 100 ext. Regards, Luis Morales On Mon, Mar 9, 2009 at 8:01 AM, Elliot Murdock murdo...@gmail.com wrote: Hello Everybody! I am currently setting up an Asterisk server for medium to high load (approximately 20-35 concurrent phone lines). Do you think the following specs will sufficiently satisfy this system? CPU: XeonQC3220 2.4GHZ 8M RAM: 2X2GB/800 Harddrive: 1X250GB I could add harddrives and partition them into /var and /log directories to help with diskdrive throughput. Thanks! Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Setup Advice
Elliot Murdock wrote: Hello Everybody! I am currently setting up an Asterisk server for medium to high load (approximately 20-35 concurrent phone lines). Do you think the following specs will sufficiently satisfy this system? CPU: XeonQC3220 2.4GHZ 8M RAM: 2X2GB/800 Harddrive: 1X250GB I could add harddrives and partition them into /var and /log directories to help with diskdrive throughput. Thanks! Elliot I'm sure this is common sense, but make sure you have a plan B for when that HD fails. It will. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: add a new queue strategy: SBR
Hi., do you think that sbr policy in queue strategy will be useful? Bye -- Forwarded message -- From: nik600 nik...@gmail.com Date: Sat, 7 Mar 2009 15:21:14 +0100 Subject: add a new queue strategy: SBR To: Asterisk Developers Mailing List asterisk-...@lists.digium.com Hi to all isn't there any plan to add the Skills Based Routing strategy in queues.conf? I think that it will be enough to add an int skill to the struct member and then order the member by skill desc. Is it enough to add this type of strategy in calc_metric in app_queue.c ? thanks -- /*/ nik600 http://www.kumbe.it -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing success rate on PRI
My recommendation would be to stay away from VoIP even T.38 whenever possible. That said yout best option is to use TDM, for that you can either use 1 single span T/E1 from digium and an analog TDM card for FXS. Or you could uee a dual span T/E1 card and a channel bank with FXS ports. While the later is more expensive I prefer that, since it gives lots more options in the long run. On 3/8/09, Marco marcota...@libero.it wrote: Hi List, I've been using PSTN-ATA + Asterisk + IAXModem + Hylafax since three years on my lab test setup and I appreciate it. Moreover the global quantity of fax handled by this setup is not very high. I'll be involved in a more complex system for a customer and I would like to ask to All of you if you have experiences and/or statistical results on faxing success and failure rate. The system I have to deploy will operate in the following context: - It will be interfaced to an E1 PRI - It will be able to send and receive faxes (by e-mail and/or virtual printers) - It will be able to send faxes from a normal fax machine. The system will be placed on the same building, i.e. only private ethernet trunks. I'm thinking to this type of solution: - Patton external unit for E1 - Asterisk 1.4 + IAXModem + Hylafax - An external ATA for the fax machine but I'm open to any other possible solution (I'm thinking to have a demodulation on Patton and talk T38 with Asterisk 1.6). The fax volume will be high because actually the customer has a ZFax software system with 12 fax-modem installed (that will be replaced by the system). I know that this was already asked in this list in the past, but I would like to know if someone has experience on this and could share their opinion, tricks and/or statistical results on failure/success rate when faxing. I think that this could be useful to other people have to realize a system like that one depicted. Thank you in advance. Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing success rate on PRI
When I said to stay away from VoIP I meant when it comes to faxing. On 3/8/09, C F shma...@gmail.com wrote: My recommendation would be to stay away from VoIP even T.38 whenever possible. That said yout best option is to use TDM, for that you can either use 1 single span T/E1 from digium and an analog TDM card for FXS. Or you could uee a dual span T/E1 card and a channel bank with FXS ports. While the later is more expensive I prefer that, since it gives lots more options in the long run. On 3/8/09, Marco marcota...@libero.it wrote: Hi List, I've been using PSTN-ATA + Asterisk + IAXModem + Hylafax since three years on my lab test setup and I appreciate it. Moreover the global quantity of fax handled by this setup is not very high. I'll be involved in a more complex system for a customer and I would like to ask to All of you if you have experiences and/or statistical results on faxing success and failure rate. The system I have to deploy will operate in the following context: - It will be interfaced to an E1 PRI - It will be able to send and receive faxes (by e-mail and/or virtual printers) - It will be able to send faxes from a normal fax machine. The system will be placed on the same building, i.e. only private ethernet trunks. I'm thinking to this type of solution: - Patton external unit for E1 - Asterisk 1.4 + IAXModem + Hylafax - An external ATA for the fax machine but I'm open to any other possible solution (I'm thinking to have a demodulation on Patton and talk T38 with Asterisk 1.6). The fax volume will be high because actually the customer has a ZFax software system with 12 fax-modem installed (that will be replaced by the system). I know that this was already asked in this list in the past, but I would like to know if someone has experience on this and could share their opinion, tricks and/or statistical results on failure/success rate when faxing. I think that this could be useful to other people have to realize a system like that one depicted. Thank you in advance. Marco Signorini === INGEGNI Tech S.r.l. http://www.ingegnitech.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: add a new queue strategy: SBR
the queue already have prioritys. David 2009/3/8 nik600 nik...@gmail.com Hi., do you think that sbr policy in queue strategy will be useful? Bye -- Forwarded message -- From: nik600 nik...@gmail.com Date: Sat, 7 Mar 2009 15:21:14 +0100 Subject: add a new queue strategy: SBR To: Asterisk Developers Mailing List asterisk-...@lists.digium.com Hi to all isn't there any plan to add the Skills Based Routing strategy in queues.conf? I think that it will be enough to add an int skill to the struct member and then order the member by skill desc. Is it enough to add this type of strategy in calc_metric in app_queue.c ? thanks -- /*/ nik600 http://www.kumbe.it -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: add a new queue strategy: SBR
but priority are se to the call, not to the agent! or am i wrong? On Sun, Mar 8, 2009 at 5:32 PM, David fire ddf...@gmail.com wrote: the queue already have prioritys. David 2009/3/8 nik600 nik...@gmail.com Hi., do you think that sbr policy in queue strategy will be useful? Bye -- Forwarded message -- From: nik600 nik...@gmail.com Date: Sat, 7 Mar 2009 15:21:14 +0100 Subject: add a new queue strategy: SBR To: Asterisk Developers Mailing List asterisk-...@lists.digium.com Hi to all isn't there any plan to add the Skills Based Routing strategy in queues.conf? I think that it will be enough to add an int skill to the struct member and then order the member by skill desc. Is it enough to add this type of strategy in calc_metric in app_queue.c ? thanks -- /*/ nik600 http://www.kumbe.it -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Setup Advice
Hello! Oh, yes, I will be mirroring the harddrives in case of any failures. What is your opinion about using (software) RAID? Do you think the overhead impacts performance too much? In an ideal situation, I would use hardware RAID, but that is not feasible right now. Thanks, Elliot On Sun, Mar 8, 2009 at 4:26 PM, Jay Milk ast-us...@skimmilk.net wrote: Elliot Murdock wrote: Hello Everybody! I am currently setting up an Asterisk server for medium to high load (approximately 20-35 concurrent phone lines). Do you think the following specs will sufficiently satisfy this system? CPU: XeonQC3220 2.4GHZ 8M RAM: 2X2GB/800 Harddrive: 1X250GB I could add harddrives and partition them into /var and /log directories to help with diskdrive throughput. Thanks! Elliot I'm sure this is common sense, but make sure you have a plan B for when that HD fails. It will. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Setup Advice
On 8 Mar 2009, at 17:04, Elliot Murdock wrote: What is your opinion about using (software) RAID? Do you think the overhead impacts performance too much? There *should* be very little disk access if you get it right. We run plenty of voice stuff on software raid. Wouldn't worry. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Setup Advice
On Sun, 8 Mar 2009, Elliot Murdock wrote: Hello! Oh, yes, I will be mirroring the harddrives in case of any failures. What is your opinion about using (software) RAID? Do you think the overhead impacts performance too much? In an ideal situation, I would use hardware RAID, but that is not feasible right now. I've used Linux software RAID for over 10 years now. for me, it's my first choice. You shouldn't be doing many disk writes though - unless you're recording all calls or handling a vast amount of voicemail. And with modern hardware there shouldn't be issues that we had in the bad old days - DMA, PIO, etc. There is a double on resources required to write a block to a software RAID-1 (mirror) unit but in a modern system, you're not going to notice it. And FWIW: I regularly have systems with 20-40 extensions running on a 1GHz VIA processor, so CPU wise, you've got more than enough - unless you're transcoding Gordon Thanks, Elliot On Sun, Mar 8, 2009 at 4:26 PM, Jay Milk ast-us...@skimmilk.net wrote: Elliot Murdock wrote: Hello Everybody! I am currently setting up an Asterisk server for medium to high load (approximately 20-35 concurrent phone lines). Do you think the following specs will sufficiently satisfy this system? CPU: XeonQC3220 2.4GHZ 8M RAM: 2X2GB/800 Harddrive: 1X250GB I could add harddrives and partition them into /var and /log directories to help with diskdrive throughput. Thanks! Elliot I'm sure this is common sense, but make sure you have a plan B for when that HD fails. It will. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing success rate on PRI
Faxing over IAX locally works fine. -- Sent from mobile device On Mar 8, 2009, at 8:46 AM, Doug Lytle supp...@drdos.info wrote: benoit wrote: Remco Barendse a écrit : On Sun, 8 Mar 2009, benoit wrote: Here is my current setup: E1 = [Asterisk with TE220p] = IAX Trunk (routed network) = [Asterisk with TDM800p] = Fax/Copy Machine You'll find that faxing over IAX is problematic at best. If this is your only option, I'd suggest that you have HylaFAX+ running at both ends. Capture the faxes coming off of the E1 as PDF, do a remote send to the other fax server and have the remote HylaFAX+ do the faxing to the fax/copy machine. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple Meetme Question
Just transfer them to your meetme extension after you've called them. Just like you would transfer someone who has called you. * will then put them into that conference. Thanks. On 08/03/2009, Sven Geggus use...@fuchsschwanzdomain.de wrote: Hello, setting up Meetme was very easy. I jut added the MeetMe Application to an internal extension to be reachable by SIP and to an external extension to be reachable by ISDN. What I don't understand however is how to call somebody and drop him to the conference? I'm using Asterisk 1.4 from Debian lenny Sven -- In the land of the brave and the free, we defend our freedom with the GNU GPL (Richard M. Stallman on www.gnu.org) /me is gig...@ircnet, http://sven.gegg.us/ on the Web ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Setup Advice
Hello! There will be disk writing in these areas: 1. Logs 2. CDRs 3. MYSQL Call logs 4. Faxes and voicemail Also, there will be a lot of codec encoding/decoding from/to the PRI devices, which is my main concern with CPU load. Cheers, Elliot On Sun, Mar 8, 2009 at 7:56 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Sun, 8 Mar 2009, Elliot Murdock wrote: Hello! Oh, yes, I will be mirroring the harddrives in case of any failures. What is your opinion about using (software) RAID? Do you think the overhead impacts performance too much? In an ideal situation, I would use hardware RAID, but that is not feasible right now. I've used Linux software RAID for over 10 years now. for me, it's my first choice. You shouldn't be doing many disk writes though - unless you're recording all calls or handling a vast amount of voicemail. And with modern hardware there shouldn't be issues that we had in the bad old days - DMA, PIO, etc. There is a double on resources required to write a block to a software RAID-1 (mirror) unit but in a modern system, you're not going to notice it. And FWIW: I regularly have systems with 20-40 extensions running on a 1GHz VIA processor, so CPU wise, you've got more than enough - unless you're transcoding Gordon Thanks, Elliot On Sun, Mar 8, 2009 at 4:26 PM, Jay Milk ast-us...@skimmilk.net wrote: Elliot Murdock wrote: Hello Everybody! I am currently setting up an Asterisk server for medium to high load (approximately 20-35 concurrent phone lines). Do you think the following specs will sufficiently satisfy this system? CPU: XeonQC3220 2.4GHZ 8M RAM: 2X2GB/800 Harddrive: 1X250GB I could add harddrives and partition them into /var and /log directories to help with diskdrive throughput. Thanks! Elliot I'm sure this is common sense, but make sure you have a plan B for when that HD fails. It will. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Setup Advice
On Sun, 8 Mar 2009, Elliot Murdock wrote: Hello! There will be disk writing in these areas: 1. Logs 2. CDRs 3. MYSQL Call logs 4. Faxes and voicemail I'd not consider these to be a heavy load myself... Also, there will be a lot of codec encoding/decoding from/to the PRI devices, which is my main concern with CPU load. Why are you transcoding? Are your extension users remote? If you set ulaw or alaw to be the codec (depending on country) used by the extensions there won't by any transcoding at all. Gordon Cheers, Elliot On Sun, Mar 8, 2009 at 7:56 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Sun, 8 Mar 2009, Elliot Murdock wrote: Hello! Oh, yes, I will be mirroring the harddrives in case of any failures. What is your opinion about using (software) RAID? Do you think the overhead impacts performance too much? In an ideal situation, I would use hardware RAID, but that is not feasible right now. I've used Linux software RAID for over 10 years now. for me, it's my first choice. You shouldn't be doing many disk writes though - unless you're recording all calls or handling a vast amount of voicemail. And with modern hardware there shouldn't be issues that we had in the bad old days - DMA, PIO, etc. There is a double on resources required to write a block to a software RAID-1 (mirror) unit but in a modern system, you're not going to notice it. And FWIW: I regularly have systems with 20-40 extensions running on a 1GHz VIA processor, so CPU wise, you've got more than enough - unless you're transcoding Gordon Thanks, Elliot On Sun, Mar 8, 2009 at 4:26 PM, Jay Milk ast-us...@skimmilk.net wrote: Elliot Murdock wrote: Hello Everybody! I am currently setting up an Asterisk server for medium to high load (approximately 20-35 concurrent phone lines). Do you think the following specs will sufficiently satisfy this system? CPU: XeonQC3220 2.4GHZ 8M RAM: 2X2GB/800 Harddrive: 1X250GB I could add harddrives and partition them into /var and /log directories to help with diskdrive throughput. Thanks! Elliot I'm sure this is common sense, but make sure you have a plan B for when that HD fails. It will. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Setup Advice
Hello Gordon, Aside from alaw and ulaw, we also use G729. I am not that familiar as to how Asterisk converts PRI signals into coded format, but why wouldn't any transcoding be necessary for alaw and ulaw codecs? Regards, Elliot On Sun, Mar 8, 2009 at 8:52 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Sun, 8 Mar 2009, Elliot Murdock wrote: Hello! There will be disk writing in these areas: 1. Logs 2. CDRs 3. MYSQL Call logs 4. Faxes and voicemail I'd not consider these to be a heavy load myself... Also, there will be a lot of codec encoding/decoding from/to the PRI devices, which is my main concern with CPU load. Why are you transcoding? Are your extension users remote? If you set ulaw or alaw to be the codec (depending on country) used by the extensions there won't by any transcoding at all. Gordon Cheers, Elliot On Sun, Mar 8, 2009 at 7:56 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Sun, 8 Mar 2009, Elliot Murdock wrote: Hello! Oh, yes, I will be mirroring the harddrives in case of any failures. What is your opinion about using (software) RAID? Do you think the overhead impacts performance too much? In an ideal situation, I would use hardware RAID, but that is not feasible right now. I've used Linux software RAID for over 10 years now. for me, it's my first choice. You shouldn't be doing many disk writes though - unless you're recording all calls or handling a vast amount of voicemail. And with modern hardware there shouldn't be issues that we had in the bad old days - DMA, PIO, etc. There is a double on resources required to write a block to a software RAID-1 (mirror) unit but in a modern system, you're not going to notice it. And FWIW: I regularly have systems with 20-40 extensions running on a 1GHz VIA processor, so CPU wise, you've got more than enough - unless you're transcoding Gordon Thanks, Elliot On Sun, Mar 8, 2009 at 4:26 PM, Jay Milk ast-us...@skimmilk.net wrote: Elliot Murdock wrote: Hello Everybody! I am currently setting up an Asterisk server for medium to high load (approximately 20-35 concurrent phone lines). Do you think the following specs will sufficiently satisfy this system? CPU: XeonQC3220 2.4GHZ 8M RAM: 2X2GB/800 Harddrive: 1X250GB I could add harddrives and partition them into /var and /log directories to help with diskdrive throughput. Thanks! Elliot I'm sure this is common sense, but make sure you have a plan B for when that HD fails. It will. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Communicator with Asterisk/Trixbox
Sorry, yes it is a Windows Application. I'm running on version 7.0. I have noted and searched and found people with version 2.0 has been able to get it working. But getting in touch with those guys is a mission and a half. Thanks, D On 8/03/09 9:53 PM, Andrew Joakimsen joakim...@gmail.com wrote: Is it the Windows software, or other? I noticed the Nokia E71 mobile has an option for Cisco IP Communicator (besides the built-in SIP client) On Wed, Mar 4, 2009 at 22:32, Dorien K. Takeshi dorien.take...@webhad.co.nz wrote: Hi guys, Has anyone had any luck with getting the Cisco IP Communicator working with your Asterisk or primarily, Trixbox installation? I've tried searching the net for information, and found someone said to set it up like the 7970 hard phone, which I have, and I'm just running into the problems with it saying Error Verifying Config Info. Any and all help is appreciated. Dorien ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: add a new queue strategy: SBR
you are wrong. when you set up an agent in a queue you can put a priority. David 2009/3/8 nik600 nik...@gmail.com but priority are se to the call, not to the agent! or am i wrong? On Sun, Mar 8, 2009 at 5:32 PM, David fire ddf...@gmail.com wrote: the queue already have prioritys. David 2009/3/8 nik600 nik...@gmail.com Hi., do you think that sbr policy in queue strategy will be useful? Bye -- Forwarded message -- From: nik600 nik...@gmail.com Date: Sat, 7 Mar 2009 15:21:14 +0100 Subject: add a new queue strategy: SBR To: Asterisk Developers Mailing List asterisk-...@lists.digium.com Hi to all isn't there any plan to add the Skills Based Routing strategy in queues.conf? I think that it will be enough to add an int skill to the struct member and then order the member by skill desc. Is it enough to add this type of strategy in calc_metric in app_queue.c ? thanks -- /*/ nik600 http://www.kumbe.it -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Setup Advice
On Sun, 8 Mar 2009, Elliot Murdock wrote: Hello Gordon, Aside from alaw and ulaw, we also use G729. I am not that familiar as to how Asterisk converts PRI signals into coded format, but why wouldn't any transcoding be necessary for alaw and ulaw codecs? It shouldn't have to convert them as they'll come in in ulaw or alaw already (depending on your country: ulaw for the US and Japan I think, alaw for most other places) I guess you're using g729 for remote connections though. Always use ulaw or alaw for local/LAN connections when talking to a PRI. There was a paper published some time back which demonstrated 14 (I think) transcodes to g729 on a 1GHz processor. My own experience is that I can do 10 on simialr hardware without getting near 100% CPU, so a faster CPU with better MMX, SSE instructions, etc. ougut to do many more. Gordon Regards, Elliot On Sun, Mar 8, 2009 at 8:52 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Sun, 8 Mar 2009, Elliot Murdock wrote: Hello! There will be disk writing in these areas: 1. Logs 2. CDRs 3. MYSQL Call logs 4. Faxes and voicemail I'd not consider these to be a heavy load myself... Also, there will be a lot of codec encoding/decoding from/to the PRI devices, which is my main concern with CPU load. Why are you transcoding? Are your extension users remote? If you set ulaw or alaw to be the codec (depending on country) used by the extensions there won't by any transcoding at all. Gordon Cheers, Elliot On Sun, Mar 8, 2009 at 7:56 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Sun, 8 Mar 2009, Elliot Murdock wrote: Hello! Oh, yes, I will be mirroring the harddrives in case of any failures. What is your opinion about using (software) RAID? Do you think the overhead impacts performance too much? In an ideal situation, I would use hardware RAID, but that is not feasible right now. I've used Linux software RAID for over 10 years now. for me, it's my first choice. You shouldn't be doing many disk writes though - unless you're recording all calls or handling a vast amount of voicemail. And with modern hardware there shouldn't be issues that we had in the bad old days - DMA, PIO, etc. There is a double on resources required to write a block to a software RAID-1 (mirror) unit but in a modern system, you're not going to notice it. And FWIW: I regularly have systems with 20-40 extensions running on a 1GHz VIA processor, so CPU wise, you've got more than enough - unless you're transcoding Gordon Thanks, Elliot On Sun, Mar 8, 2009 at 4:26 PM, Jay Milk ast-us...@skimmilk.net wrote: Elliot Murdock wrote: Hello Everybody! I am currently setting up an Asterisk server for medium to high load (approximately 20-35 concurrent phone lines). Do you think the following specs will sufficiently satisfy this system? CPU: XeonQC3220 2.4GHZ 8M RAM: 2X2GB/800 Harddrive: 1X250GB I could add harddrives and partition them into /var and /log directories to help with diskdrive throughput. Thanks! Elliot I'm sure this is common sense, but make sure you have a plan B for when that HD fails. It will. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outlook integration?
Noojeeclick? http://www.noojee.com.au/Page/NoojeeClick ADM? (asterisk desktop manager?) PaulH Alan Lord (News) wrote: Dean Collins wrote: ADA Forums: http://forums.digium.com/index.php?c=8 ADA Download: http://dl1.digium.com/ADA/ADAInstall.exe ADA Administrators Guide: http://dl1.digium.com/ADA1.1/ADA_Admin_Manual.pdf Thanks for the links. I hadn't seen that before. The product is kind of interesting, but does anyone know of something similar for non-windows desktops? Thanks Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Server Setup Advice
On 9/03/2009 8:08 a.m., Elliot Murdock wrote: Hello Gordon, Aside from alaw and ulaw, we also use G729. I am not that familiar as to how Asterisk converts PRI signals into coded format, but why wouldn't any transcoding be necessary for alaw and ulaw codecs? Because a T1 uses mulaw and an E1 uses alaw (except in some E1 connections in Japan - which use ulaw). If your phones are also using ulaw then the signal would not need to be transcoded - there's really very little reason to use anything other that alaw/ulaw in a LAN situation - the only reason you would want to use G729 etc is if you have bandwidth issues. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple Meetme Question
Gavin Henry gavin.he...@gmail.com wrote: Just transfer them to your meetme extension after you've called them. Hm, how would I do this? Until now call switching usually ended for me when the call has been established. I'm using a SIP phone connected to an asterisk box which is connected to the world via various ways (ISDN, SIP, IAX2). So what would I do on the my SIP phone after the call has been established and what needs to be changed in the dialplan to actually reconnect the current call to the MeetMe Conference then? Sven -- The main thing to note is that when you choose open source you don't get a Windows operating system. (from http://www.dell.com/ubuntu) /me is gig...@ircnet, http://sven.gegg.us/ on the Web ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: add a new queue strategy: SBR
David fire wrote: you are wrong. when you set up an agent in a queue you can put a priority. David The term used in Asterisk for a queue member's priority is the word penalty. When you set up a member in queues.conf, the penalty is the third option for a member. Here's an example: member = SIP/2000,Mark Michelson,3 In the above example, Mark Michelson is the name of a queue member who can be reached by calling the interface SIP/2000. His penalty is 3. The rule for penalties is that members with lower penalties are called before members with higher penalties. If all the members of the lowest penalty are unavailable (i.e. not logged in or currently on a call) then the Queue application will attempt to call a member with a higher penalty. Caution: One shortcoming of queue member penalties is that they are not taken into account if a queue member of a low penalty does not answer a call. Say for instance that the queue application determines that there are two members available to answer an incoming call. One member has penalty 1 and the other has penalty 2. If the member with penalty 1 does not answer the call, the queue application still considers that member to be available the next time that it tries to reach a member. The member with penalty 2 will only be tried if the queue application can determine *before the call is placed* that the member with penalty 1 is unavailable. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring asterisk to revert call back to forwarder if exten is busy
I manged to get something working but It's only working when a
grandstream ip phones is the one tranfering calls. With linksys IP
phones I get a busy yone
I edited the extensions.conf
last lines of [macro-exten-vm]:
; Extensions with no Voicemail box reporting BUSY come here
exten = s-BUSY,1,NoOp(Extension is reporting BUSY and not passing to Voicemail)
; This should recover failed transfers.
exten = s-BUSY,n,GotoIf($[${LEN(${BLINDTRANSFER})}
0]?custom-MANAGE_LOST_TRANSFERS,s,1)
exten = s-BUSY,n,Playtones(busy)
exten = s-BUSY,n,Busy(20)
; Anything but BUSY comes here
; This should recover failed transfers.
exten = _s-.,1,GotoIf($[${LEN(${BLINDTRANSFER})}
0]?custom-MANAGE_LOST_TRANSFERS,s,1)
exten = _s-.,n,Playtones(congestion)
exten = _s-.,n,Congestion(10)
and added a new context on extensions_custom.conf
[custom-MANAGE_LOST_TRANSFERS]
exten = s,1,Answer()
exten = s,n,Playback(please-wait-bouncing-back)
; Supposing there are 4-digit extensions here - no error checking
exten = s,n,Set(RETURN_EXT=${BLINDTRANSFER:4:3})
exten = s,n,Goto(from-internal,${RETURN_EXT},1)
exten = s,n,HangUp()
What could be the problem with linksys phones?
_
Paul Hales wrote:
Can I assume that you want this only for blind transfers?
I have done this previously, but I lost my copy of the work (and it was
a proof of concept only)
It involved the ${BLINDTRANSFER} variable, which catches the number that
made the blind transfer and making macro-stdexten (or your equivalent)
dial that variable in the case of the dial status being treated as BUSY.
To get a 'busy' will involve single line phones, or disabling call
waiting on the phone receiving the call.
regards,
PaulH
James Mutuku wrote:
Hellos,
I want to configure asterisk so that if exten A transfers a call to
exten B, and B is either busy or the call is not answered, the call
returns back to A. Is this possible?
Please help
James
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begin:vcard
fn:James Mutuku
n:Mutuku;James
org:Agile Systems Limited;Technical Department
adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya
email;internet:jnmut...@gmail.com,jmut...@agile.co.ke
title:Lead Consultant
tel;work:+254-722-490994
tel;home:+254-722-490994
tel;cell:+254-722-490994
url:www.agile.co.ke
version:2.1
end:vcard
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