Re: [asterisk-users] Strange behaviour with Dial cmd
On 2/14/07, Yuan LIU [EMAIL PROTECTED] wrote: Well, you'll have to decide how you want to hang up the caller: Do you want him/her to be ignored, or to be told that you are not available (like an answering machine)? You also need to tell Asterisk how to determine if the next invite comes from the same caller during the same session. (These are not very easy tasks but doable.) In either case, you need to add a flag to your dial plan, set it after it rings your cell, and reset it once Asterisk determines that THIS caller has been hung up. (Of course you can do what Vacation does in E-mail: set up a flag for each identifiable caller, and only call your mobile once until you reset them all. The algorithm would be simpler but more unidentifiable callers will be ignored.) Your dial plan will check this flag before ringing your cell, then branch accordingly. Hope this helps. Yuan Liu Thank you for the answer. Right know I solved sending after everything to the voicemail. But I tought that hangup was suppose to close the call, however, is not the case and a really did not catch why. I will try to read a bit more. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] directed call pickup with PICKUPMARK
Hi! i have a problem with the PICKUPMARK of the Pickup() application. E.g. A calls B. B is ringing. C wants to pickup B. To make this work with PICKUPMARK I have to add the variable PICKUPMARK to B. But how can I do this? B is just created inside the Dial() application. thanks klaus PS: If you wonder why I use the PICKUPMARK - Pickup(exten) does not work for me as the incoming calls will be handled in macros and thus the extension of the ringing call is 's' and not the dialed extension. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best FXO Gateway
On Thu, 15 Feb 2007, jameson asterisk wrote: I'm currently looking to deploy an Asterisk server using an FXO media gateway to connect to the PSTN and was looking for any user experiences that may aid in selecting a gateway. Specifically i'm looking for a 4-port model under 500 dollars. Within this category exists: MediaTrix 1204 Grandstream GXW-4104 AudioCodes MP114 I've read over Voip-info.org regarding these products and numerous mentions of problematic setup and configuration seems to be a common theme. Can anyone provide a recommendation based on user experience? Feel free to suggest an alternative gateway if one stands out. I've no experience of anything like this other than experiments with a Grandtream HT-488 (one port), but I'm curious as to why you want an external box to do the work rather than put a PCI card inside your asterisk box (eg. Digium TDM400, Sangoma, etc.) Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] moving WiFi phone
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 15 Feb 2007, at 01:39, Leo Ann Boon wrote: Bruce Reeves wrote: In my experience having ap's with the same SSID and 3 channels of separation overlapping worked if the phone could roam. Recommended is 5 channels of separation. Ronald, Just be aware that even if the phone supports AP roaming, there's no guarantee that the call will continue smoothly from AP to AP. In some cases, it might take a few seconds to handover. I have two APs (Apple AirPorts) sending on the _same_ channel. Handover works perfect with no discernible loss of connectivity or audio using a Siemens SL75. The handover cannot even be noticed. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (Darwin) iD8DBQFF1CK+RAx5nvEhZLIRAgJgAJ9rXMM7xQuQNaYCdUSziFz0UVbE4ACfdSuH FeEFtrmJttLNUBdrIi8DuTU= =jTGX -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
- Original Message - From: Nic Bellamy [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 13, 2007 8:40 PM Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC) Larry Shields wrote: I recently read about the following new technologies from Digium. Has anyone tried the new HPEC or knows when it will be available? It's out now, and I've tried it - the difference between HPEC and MG2 from trunk is stunning - in situations with bad echo where MG2 can take ten or more seconds to converge to a reasonable degree, HPEC does it in perhaps 300ms - converging on my intake of breath before I say hello, and absolutely no echo after that unless I purposefully go out of my way to screw it up (whistling/blowing into the handpiece for instance - even then, the malfunction is minimal). You can now buy it from the Digium website (US$10 per channel), or if you have an in-warranty Digium card, email through the serial numbers to Digium support and they'll give you a key (this is what I did). You'll need Zaptel 1.2.13 to make it go. It does take quite a bit of CPU though - perhaps 70% more compared to MG2-trunk for the same number of taps from my rough measurements. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Does anyone know if the HPEC will work on a Sangoma A200 / 2 port FXO card? (I'm assuming so as it still uses Zapel) I've 2 PSTN lines one of which I cannot get rid of the echo, I've tried a 2GHz machine as apposed to my normal P3 650MHz and this made no difference. Would the 650Mhz be enough to run HPEC on one line (I assume only needing one licence) This is what Digium say on their web site: Digium recommends that users requiring 8 channels at 1024 taps run a PC comparible to a 3.0 GHz Pentium 4, while users only requiring 4 channels at 1024 taps may run a 2.5 GHz Pentium Celeron. The CPU requirements are such that it is impractical to operate this echo canceller at 1024 taps for a full T1 or E1 of channels. Many Thanks Harvey ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] moving WiFi phone
Jens Vagelpohl wrote: I have two APs (Apple AirPorts) sending on the _same_ channel. Handover works perfect with no discernible loss of connectivity or audio using a Siemens SL75. The handover cannot even be noticed. as I know, best practice says, that neighboring AP should use _non overlapping_ channels... :-\ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Symbian IAX client
Hi all, Does anyone know of an IAX client for Symbian? I have an e61 and would like to make calls through my home Asterisk box from places where I have WiFi access, as NAT is in the way I suspect that it'll be a pain to get SIP working like that as the NAT router doesn't do SIP connection tracking. Thanks, Peter. This message has been comprehensively scanned for viruses, please visit http://www.avg.power.net.uk/ for details. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] moving WiFi phone
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 15 Feb 2007, at 10:23, Pavel Jezek wrote: Jens Vagelpohl wrote: I have two APs (Apple AirPorts) sending on the _same_ channel. Handover works perfect with no discernible loss of connectivity or audio using a Siemens SL75. The handover cannot even be noticed. as I know, best practice says, that neighboring AP should use _non overlapping_ channels... :-\ works for me is all I can say. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (Darwin) iD8DBQFF1C04RAx5nvEhZLIRAtuKAJ94ZKW0/WZkPnoM9hUQm+hHAJ+5cACgtir5 1fRums89u32Kleaf0fCuP+Y= =IFN9 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] moving WiFi phone
Pavel Jezek ha scritto: Jens Vagelpohl wrote: I have two APs (Apple AirPorts) sending on the _same_ channel. Handover works perfect with no discernible loss of connectivity or audio using a Siemens SL75. The handover cannot even be noticed. as I know, best practice says, that neighboring AP should use _non overlapping_ channels... :-\ After *months* of troubles using a 14 APs network with same SSID, WPA/TKIP security model, tx power settings and channels carefully distributed in order to be as non overlapping as possible, including a controller capable of performing fast layer2 reauthentication (e.g. something like caching WPA keys between access points), I always got VERY POOR roaming performance. I've tested these phones: UTStarcom F1000 UTStarcom F1000g UTStarcom F3000g Siemens Gigaset SL75 WLAN Nokia E60 Nokia E70 Samsung WIP6000 Linksys WIP300 I was desperate. I took a bold step. I downgraded to WEP-128 (I know it's weak) and, despite the recommendations from any good wifi networking guide, I SET ALL APs ON THE SAME CHANNEL. Don't ask me why, but now roaming is PERFECT, never had a call dropped or even a hiss or crackling noise during conversation. I can even run or move over the site hangar on forklift trucks while talking on the phone, at 15-20 mph. Luckily there are no high throughput demands for data transmission (PDAs, notebooks, etc) over the wifi network, so I didn't got performance issues. Imho roaming support on 802.11 wifi networks is far from being usable... -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange behaviour with Dial cmd
From: Il Neofita [EMAIL PROTECTED] Date: Thu, 15 Feb 2007 03:37:14 -0500 But I tought that hangup was suppose to close the call, however, is not the case and a really did not catch why. Now I see where the confusion comes from. Asterisk doesn't really speak English - or Chinese for that matter:-) In telephony, there is no way for the callee to tell the caller to stop ringing - unless you answer it first. Once you answer, you can do a number of things, the rudest being to immediately hang up. (I saw live people doing this intentionally.) Your only other option really is to ignore. I just thought up this simple method to ignore: divert the dial plan to simply Wait() an unreasonable amount of time in hope that the caller hangs up. exten = s,1,Dial(yourcell,5) exten = s,n,Wait(300) That's assuming your provider provides disconnect supervision. You can also Play(prerecorded,noanswer) if your provider supports it. (Won't hurt to try even if not - but disconnect supervision is a must.) Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax with T.38
Hi all, I make mistakes in my explanation, so I will try to re-explain my problem... I want to send fax with FoIP. Analog Fax PATTON SN4960 Asterisk PATTON M-ATA Analog Fax 2 In the Patton SN4960 configuration I have : profile voip FOIP codec 1 g729 rx-length 20 tx-length 20 no-silence-suppression codec 2 g711alaw64k rx-length 30 tx-length 30 no-silence-suppression codec 3 g711ulaw64k rx-length 30 tx-length 30 no-silence-suppression dtmf-relay signaling dejitter-max-delay 100 fax transmission 1 relay t38-udp fax redundancy low-speed 2 high-speed 1 fax detection fax-frames modem transmission 1 bypass g711alaw64k modem bypass-method nse On Patton M-ATA : 1. codec alaw 2. codec ulaw 3. codec g729 No silence suppression on these codecs. I not use this option FAX without T.38(Use G.711 fax) On asterisk side I have: [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=alaw dtmfmode = rfc2833 rtcachefriends=yes realm=vtxvoip useragent=VTX SIP rtupdate=yes language=en tos=184 notifyringing=yes t38pt_udptl=yes And t38pt_udptl=yes in the 2 PATTONs sip accounts ... Then, I made a trace when I call from Analog FAX 2 to Analog fax 1 I received T.38 packets from the Patton sn4960 but no T.38 packets go through the Asterisk And on the asterisk I have 3 WARNINGS: [Feb 15 10:05:24] WARNING[9167]: channel.c:3033 ast_channel_make_compatible: No path to translate from SIP/sip_trunk_gva-01-0071e6b0(256) to SIP/0xxx0379xx-0070a490(8) [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 What I really not understand it's why asterisk try to translate from ulaw to g729 !!! I disallow all and allow just the alaw codec ... more than this, I remove the g729 licence file ... Do you have an idea for me ?? Thanks a lot, Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queues do not accept calls if all agent are busy?
Hi, cud any one help me figuring out the problem... When the agent in a queue is engaged, it cannot accept anymore calls, below is the script; -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/2063||tr) in new stack -- Called 2063 -- Local/[EMAIL PROTECTED],1 is ringing -- Got SIP response 486 Busy Here back from 10.19.1.158 -- SIP/2063-084a6c18 is busy == Everyone is busy/congested at this time (1:1/0/0) -- Executing NoOp(Local/[EMAIL PROTECTED],2, Returned to dolocaldial with DIALSTATUS BUSY) in new stack -- Executing Macro(Local/[EMAIL PROTECTED],2, outisbusy|) in new stack -- Executing Playback(Local/[EMAIL PROTECTED],2, all-circuits-busy-now) in new stack -- Playing 'all-circuits-busy-now' (language 'en') -- Local/[EMAIL PROTECTED],1 answered SIP/10.19.1.157-084eec28 -- Stopped music on hold on SIP/10.19.1.157-084eec28 -- Executing Playback(Local/[EMAIL PROTECTED],2, pls-try-call-later) in new stack -- Playing 'pls-try-call-later' (language 'en') -- Executing Macro(Local/[EMAIL PROTECTED],2, hangupcall) in new stack Thanks Angel - Expecting? Get great news right away with email Auto-Check. Try the Yahoo! Mail Beta.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] End Wrap-up Time?
Hi James, The only solution I've managed to find so far is to set the wrap-up time to 5 seconds and tell the operators that if they need more time, they need to put themselves on pause. See PauseQueueMember and UnpauseQueueMember. If someone has a better solution, I'd be most pleased to hear of it! James Fromm wrote: Does anyone have a solution to allow an agent to selectively end his wrap-up time? We define a wrap-up time of 60 seconds to allow our agents to finish their notes from a call. In some cases, the full 60 seconds is not needed and our agents would like to end their wrap-up time. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax with T.38
Hi all, I make mistakes in my explanation, so I will try to re-explain my problem… I want to send fax with FoIP. Analog Fax ← PSTN → PATTON SN4960 ←T.38→ Asterisk ←T.38→ PATTON M-ATA ←Analog→ Analog Fax 2 In the Patton SN4960 configuration I have : profile voip FOIP codec 1 g729 rx-length 20 tx-length 20 no-silence-suppression codec 2 g711alaw64k rx-length 30 tx-length 30 no-silence-suppression codec 3 g711ulaw64k rx-length 30 tx-length 30 no-silence-suppression dtmf-relay signaling dejitter-max-delay 100 fax transmission 1 relay t38-udp fax redundancy low-speed 2 high-speed 1 fax detection fax-frames modem transmission 1 bypass g711alaw64k modem bypass-method nse On Patton M-ATA : 1. codec alaw 2. codec ulaw 3. codec g729 No silence suppression on these codecs. I not use this option “FAX without T.38(Use G.711 fax)” On asterisk side I have: [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=alaw dtmfmode = rfc2833 rtcachefriends=yes realm=vtxvoip useragent=VTX SIP rtupdate=yes language=en tos=184 notifyringing=yes t38pt_udptl=yes And t38pt_udptl=yes in the 2 PATTONs sip accounts … Then, I made a trace when I call from Analog FAX 2 to Analog fax 1 …. I received T.38 packets from the Patton sn4960 but no T.38 packets go through the Asterisk …. And on the asterisk I have 3 WARNINGS: [Feb 15 10:05:24] WARNING[9167]: channel.c:3033 ast_channel_make_compatible: No path to translate from SIP/sip_trunk_gva-01-0071e6b0(256) to SIP/0xxx0379xx-0070a490(8) [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 What I really not understand it’s why asterisk try to translate from ulaw to g729 !!! I disallow all and allow just the alaw codec … more than this, I remove the g729 licence file … Do you have an idea for me ?? Thanks a lot, Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CAS signalling on span 2 conflicts with HDLC with FCS check on channel 20
Yes I m using E1 the equivalent of T2 (31 channels) 2007/2/14, Melcon Moraes [EMAIL PROTECTED]: You should answer questions asked to you. I saw Tzafrir Cohen asking you if you were using a E1 PRI. Are you? []'s MM -Original Message- From: younss azzayani [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Sent: Wed, 14 Feb 2007 15:05:51 + Delivered: Wed, 14 Feb 2007 12:32:47 Subject:[asterisk-users] CAS signalling on span 2 conflicts with HDLC with FCS check on channel 20 hello my friends, when i make a genzaptelconf i get this message CAS signalling on span 2 conflicts with HDLC with FCS check on channel *** Any idea Please? I m installing zaptel 1.4 i checked in http://bugs.digium.com/view.php?id=7860; that it's a bug but beacause i m a newbie in asterisk i can't undrestand what exactly mean Thank You ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,1171465751.116199.25030.vacoas.hst.terra.com.br,4109,Des15,Des15 --Original Message Ends-- -- Melcon Moraes [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] genzaptool from xorcom
ok thank you Cohen thank you very much 2007/2/14, Tzafrir Cohen [EMAIL PROTECTED]: On Wed, Feb 14, 2007 at 03:44:25PM +, younss azzayani wrote: Thank You Cohen What card do you have? * Digium TE110P TDM400P, think the problem is with TE110P (configured as span 2) because i remark that the dchannel=20 * What is th output of: cat /proc/zaptel/* * Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 1 WCTDM/0/0 FXSKS 2 WCTDM/0/1 FXSKS 3 WCTDM/0/2 FXSKS 4 WCTDM/0/3 FXSKS Span 2: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3/CCS 5 WCT1/0/1 Clear 6 WCT1/0/2 Clear 7 WCT1/0/3 Clear 8 WCT1/0/4 Clear 9 WCT1/0/5 Clear 10 WCT1/0/6 Clear 11 WCT1/0/7 Clear 12 WCT1/0/8 Clear 13 WCT1/0/9 Clear 14 WCT1/0/10 Clear 15 WCT1/0/11 Clear 16 WCT1/0/12 Clear 17 WCT1/0/13 Clear 18 WCT1/0/14 Clear 19 WCT1/0/15 Clear 13 WCT1/0/9 Clear 14 WCT1/0/10 Clear 15 WCT1/0/11 Clear 16 WCT1/0/12 Clear 17 WCT1/0/13 Clear 18 WCT1/0/14 Clear 19 WCT1/0/15 Clear 20 WCT1/0/16 This is the D channel, right? Is the connection a E1 PRI? 21 WCT1/0/17 22 WCT1/0/18 23 WCT1/0/19 24 WCT1/0/20 25 WCT1/0/21 26 WCT1/0/22 27 WCT1/0/23 28 WCT1/0/24 29 WCT1/0/25 30 WCT1/0/26 31 WCT1/0/27 32 WCT1/0/28 33 WCT1/0/29 34 WCT1/0/30 35 WCT1/0/31 31 channels, as expected. What is the generated /etc/zaptel.conf ? ** # Autogenerated by ./genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 fxsks=1 fxsks=2 fxsks=3 fxsks=4 # Span 2: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3/CCS span=2,1,1,ccs,hdb3 bchan=5-19,21-35 dchan=20 # Global data loadzone= us defaultzone = us The error you get is from a place in ztcfg's code that applies some sanity checks to the signalling it sends to channel no. 16 of a span. If they are not met, that channel cannot be considered a D channel. I didn't understand those conditions exactly. In one specific case were I helped someone on #asterisk that guy eventually removed the sanity check from ztcfg and moved on. Whether or not this is a wise thing to do, I don't know. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel Won't Compile In Slackware 11.0 Kernel 2.4.33
Hi, Even a default Slackware 11.0 with 2.4.33 kernel source failed to look for page-flags.h and I do not have a card that your are referring to because this is a development machine on a laptop. It used to work before but the current source tree which i get into a week ago started to break out for my case. Hi Demuel, 1st. Do you have a card that support 'zttranscode' ? if Yes, go ahead with this, if no, there's no use for you to be compiling it. 2nd. Do NOT do this: [EMAIL PROTECTED]:/usr/src/linux/include/linux$ sudo make updatedb Do this [EMAIL PROTECTED]:/usr/src/linux/include/linux$ updatedb (as root, or you will have to permit that command 'updatedb' in the sudoers list for the user 'demuel', in ur case) then: [EMAIL PROTECTED]:/usr/src/linux/include/linux$ slocate page-flags.h If u dont have it (slocate will certainly finds it if u do), then try to get it (of course, not just that file cause you could be missing another one in the farther process of compilation). Try to find out of what package or source where that file belongs to, and get it... J. Espinal [EMAIL PROTECTED] wrote: [EMAIL PROTECTED]:/usr/src/linux/include/linux$ pwd /usr/src/linux/include/linux [EMAIL PROTECTED]:/usr/src/linux/include/linux$ sudo make updatedb make: *** No rule to make target `updatedb'. Stop. [EMAIL PROTECTED]:/usr/src/linux/include/linux$ ls -la page-flags.h /bin/ls: page-flags.h: No such file or directory [EMAIL PROTECTED]:/usr/src/linux/include/linux$ Did i missed something down here? Weird thing is, even a fresh install of slackware produced the same kind of error. Actually, it used to be working about a week before I made a source upgrade. Any thoughts? Regards, Demuel On Tue, Feb 13, 2007 at 02:23:21PM +, J. Espinal wrote: make a 'updatedb' , and look for 'page-flags.h' , i think that you might be missing that file, under the include/ directory in the linux kernel source directory. J. Espinal, [EMAIL PROTECTED] wrote: Anybody, I have download asterisk 1.4 via svn. whem I compiled it, I got the following error: /lib/modules/2.4.33.3/build/include/asm/system.h:190: warning: dereferencing type-punned pointer will break strict-aliasing rules zttranscode.c:37:30: linux/page-flags.h: No such file or directory make[1]: *** [zttranscode.o] Error 1 make[1]: Leaving directory `/home/kingkong/code/projects/asterisk/source/zaptel-1.4' make: *** [all] Error 2 make a 'updatedb' , and look for 'page-flags.h' , i think that you might be missing that file, under the include/ directory in the linux kernel source directory. Better yet: simply don't build zttranscode, unless you have a card that actually supports it... -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Soyo G668 (IP Phone)
This is a PA-1688 chip phone. Give a look at http://www.aredfox.com/. It has what you need. Look for Pamtool. Isamar On Wed, 14 Feb 2007, Alcides Cremonezi wrote: Hi! Everyone, This IP phone came configured for to be used with Soyo VoIP service. I would like to set it up to work with my asterisk server with IAX2. I followed the procedure described on the Soyo website, but samething strange happens during the firmware actualization that makes the display half black, and the telephone did not work well. I changed the device for a new one, but before update the firmware again I would like to listen someone who would like to share experiences or give me a hint about it... Thanks in advance, Alcides ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
Wireless wrote: - Original Message - From: Nic Bellamy [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 13, 2007 8:40 PM Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC) Larry Shields wrote: I recently read about the following new technologies from Digium. Has anyone tried the new HPEC or knows when it will be available? It's out now, and I've tried it - the difference between HPEC and MG2 from trunk is stunning - in situations with bad echo where MG2 can take ten or more seconds to converge to a reasonable degree, HPEC does it in perhaps 300ms - converging on my intake of breath before I say hello, and absolutely no echo after that unless I purposefully go out of my way to screw it up (whistling/blowing into the handpiece for instance - even then, the malfunction is minimal). You can now buy it from the Digium website (US$10 per channel), or if you have an in-warranty Digium card, email through the serial numbers to Digium support and they'll give you a key (this is what I did). You'll need Zaptel 1.2.13 to make it go. It does take quite a bit of CPU though - perhaps 70% more compared to MG2-trunk for the same number of taps from my rough measurements. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Does anyone know if the HPEC will work on a Sangoma A200 / 2 port FXO card? (I'm assuming so as it still uses Zapel) I've 2 PSTN lines one of which I cannot get rid of the echo, I've tried a 2GHz machine as apposed to my normal P3 650MHz and this made no difference. Would the 650Mhz be enough to run HPEC on one line (I assume only needing one licence) This is what Digium say on their web site: Digium recommends that users requiring 8 channels at 1024 taps run a PC comparible to a 3.0 GHz Pentium 4, while users only requiring 4 channels at 1024 taps may run a 2.5 GHz Pentium Celeron. The CPU requirements are such that it is impractical to operate this echo canceller at 1024 taps for a full T1 or E1 of channels. It looks like octasic have started supplying their echo canceller as host software for zaptel now. I expect either canceller would work with the Sangoma cards, as they currently sit in the zaptel framework too. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Interruptible announcements in queue application
Hello all, Ive found another issue with the queue application. Assuming Ive configured a queue with a long periodic announcement and have two queue members assigned. Both queue members are busy at a time, while another caller is joining the queue. After a while the periodic announcement is played back to the caller, in that case it takes about 40 seconds to be played back. If then one of the two agents becomes available, the call is unfortunately not routed to the agent, until the playback of the announcement has finished. If you display the agent status and the queue to a supervisor he can see that there are callers waiting up to 40 seconds, even if there are available queue members. For inbound call centers tat is more than suboptimal. Does anybody know if somebody already created a patch to interrupt queue announcement when an agent becomes ready? Thanks and Regards Markus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call forwarding
Hi All,
I'm using asterisk 1.2.15 and call forwarding doesnt work for me.
from my extensions.conf:
; Unconditional Call Forward
exten = _*21*X.,1,NoCDR
exten = _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4})
exten = _*21*X.,3,Playback(vm-saved)
exten = _*21*X.,4,Hangup
exten = #21#,1,NoCDR
exten = #21#,2,DBdel(CFIM/${CALLERID(NUM)})
exten = #21#,3,Playback(auth-thankyou)
exten = #21#,4,Hangup
debug from asterisk CLI:
-- Executing NoCDR(SIP/dzalewski-081afaf0, ) in new stack
Feb 15 15:00:19 NOTICE[32307]: cdr.c:443 ast_cdr_free: CDR on
channel 'SIP/dzalewski-081afaf0' not posted
Feb 15 15:00:19 NOTICE[32307]: cdr.c:445 ast_cdr_free: CDR on
channel 'SIP/dzalewski-081afaf0' lacks end
-- Executing Set(SIP/dzalewski-081afaf0, DB(CFIM/200)=204) in new
stack
-- Executing Playback(SIP/dzalewski-081afaf0, vm-saved) in new stack
-- Playing 'vm-saved' (language 'en')
-- Executing Hangup(SIP/dzalewski-081afaf0, ) in new stack
== Spawn extension (from-internal, *21*204, 4) exited non-zero
on 'SIP/dzalewski-081afaf0'
Thank you in advance,
Dominik
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Re: [asterisk-users] Interruptible announcements in queue application
I am not aware of one.Why would you want your queue announcement interupted? When we had our Nortel, I found that feature annoying because people would be transfered to the agent half way through a message. Confusing. I configured it to not break out of an annoucement. On 2/15/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello all, I've found another issue with the queue application. Assuming I've configured a queue with a long periodic announcement and have two queue members assigned. Both queue members are busy at a time, while another caller is joining the queue. After a while the periodic announcement is played back to the caller, in that case it takes about 40 seconds to be played back. If then one of the two agents becomes available, the call is unfortunately not routed to the agent, until the playback of the announcement has finished. If you display the agent status and the queue to a supervisor he can see that there are callers waiting up to 40 seconds, even if there are available queue members. For inbound call centers tat is more than suboptimal. Does anybody know if somebody already created a patch to interrupt queue announcement when an agent becomes ready? Thanks and Regards Markus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call forwarding
Am 15.02.2007 um 14:06 schrieb Dominik Zalewski:
exten = _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4})
Just use ${CALLERID(num)} and not ${CALLERID(NUM)}.
Stefan
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? - http://www.das-asterisk-buch.de
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
On Thursday 15 February 2007 6:51 am, Steve Underwood wrote: It looks like octasic have started supplying their echo canceller as host software for zaptel now. I expect either canceller would work with the Sangoma cards, as they currently sit in the zaptel framework too. Out of curiosity, why do you suppose that it is the Octasic algorithm which is used in Digium's HPEC? I have no reasons to suspect otherwise, but I'm curious as to your reasons for suspecting that is indeed the case. Oh, and sorry about the incorrect attribution as to which Steve wrote and maintains spandsp. I always get yourself and Steven Critchfield mixed up. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel Won't Compile In Slackware 11.0 Kernel 2.4.33
Hi Demuel, Look, i think (im not very sure yet) that the *page-flags.h* file belongs to kernel = 2.5.x, not to the 2.4.x, Im using Slackware 10.2 , I have not upgraded yet to the 11.0 but i think that it comes with the 2.6.x kernel like a native kernel (not in /test/ directory anymore), 1. Why dont u try the 2.6.x kernel and get the kernel source ? 2. what's the result of runing this command on ur computer : *uname -a* 3. Have u try another version (not the SVN) ? [EMAIL PROTECTED] wrote: Hi, Even a default Slackware 11.0 with 2.4.33 kernel source failed to look for page-flags.h and I do not have a card that your are referring to because this is a development machine on a laptop. It used to work before but the current source tree which i get into a week ago started to break out for my case. Hi Demuel, 1st. Do you have a card that support 'zttranscode' ? if Yes, go ahead with this, if no, there's no use for you to be compiling it. 2nd. Do NOT do this: [EMAIL PROTECTED]:/usr/src/linux/include/linux$ sudo make updatedb Do this [EMAIL PROTECTED]:/usr/src/linux/include/linux$ updatedb (as root, or you will have to permit that command 'updatedb' in the sudoers list for the user 'demuel', in ur case) then: [EMAIL PROTECTED]:/usr/src/linux/include/linux$ slocate page-flags.h If u dont have it (slocate will certainly finds it if u do), then try to get it (of course, not just that file cause you could be missing another one in the farther process of compilation). Try to find out of what package or source where that file belongs to, and get it... J. Espinal [EMAIL PROTECTED] wrote: [EMAIL PROTECTED]:/usr/src/linux/include/linux$ pwd /usr/src/linux/include/linux [EMAIL PROTECTED]:/usr/src/linux/include/linux$ sudo make updatedb make: *** No rule to make target `updatedb'. Stop. [EMAIL PROTECTED]:/usr/src/linux/include/linux$ ls -la page-flags.h /bin/ls: page-flags.h: No such file or directory [EMAIL PROTECTED]:/usr/src/linux/include/linux$ Did i missed something down here? Weird thing is, even a fresh install of slackware produced the same kind of error. Actually, it used to be working about a week before I made a source upgrade. Any thoughts? Regards, Demuel On Tue, Feb 13, 2007 at 02:23:21PM +, J. Espinal wrote: make a 'updatedb' , and look for 'page-flags.h' , i think that you might be missing that file, under the include/ directory in the linux kernel source directory. J. Espinal, [EMAIL PROTECTED] wrote: Anybody, I have download asterisk 1.4 via svn. whem I compiled it, I got the following error: /lib/modules/2.4.33.3/build/include/asm/system.h:190: warning: dereferencing type-punned pointer will break strict-aliasing rules zttranscode.c:37:30: linux/page-flags.h: No such file or directory make[1]: *** [zttranscode.o] Error 1 make[1]: Leaving directory `/home/kingkong/code/projects/asterisk/source/zaptel-1.4' make: *** [all] Error 2 make a 'updatedb' , and look for 'page-flags.h' , i think that you might be missing that file, under the include/ directory in the linux kernel source directory. Better yet: simply don't build zttranscode, unless you have a card that actually supports it... -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call forwarding
On Thursday 15 February 2007 03:22:58 pm Stefan Wintermeyer wrote:
Am 15.02.2007 um 14:06 schrieb Dominik Zalewski:
exten = _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4})
Just use ${CALLERID(num)} and not ${CALLERID(NUM)}.
Stefan
it didnt help :( Is there is other way to implement call forwarding?
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Re: [asterisk-users] Call forwarding
On 2/15/07, Dominik Zalewski [EMAIL PROTECTED] wrote:
Hi All,
I'm using asterisk 1.2.15 and call forwarding doesnt work for me.
from my extensions.conf:
; Unconditional Call Forward
exten = _*21*X.,1,NoCDR
exten = _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4})
exten = _*21*X.,3,Playback(vm-saved)
exten = _*21*X.,4,Hangup
exten = #21#,1,NoCDR
exten = #21#,2,DBdel(CFIM/${CALLERID(NUM)})
exten = #21#,3,Playback(auth-thankyou)
exten = #21#,4,Hangup
debug from asterisk CLI:
-- Executing NoCDR(SIP/dzalewski-081afaf0, ) in new stack
Feb 15 15:00:19 NOTICE[32307]: cdr.c:443 ast_cdr_free: CDR on
channel 'SIP/dzalewski-081afaf0' not posted
Feb 15 15:00:19 NOTICE[32307]: cdr.c:445 ast_cdr_free: CDR on
channel 'SIP/dzalewski-081afaf0' lacks end
-- Executing Set(SIP/dzalewski-081afaf0, DB(CFIM/200)=204) in new
stack
-- Executing Playback(SIP/dzalewski-081afaf0, vm-saved) in new stack
-- Playing 'vm-saved' (language 'en')
-- Executing Hangup(SIP/dzalewski-081afaf0, ) in new stack
== Spawn extension (from-internal, *21*204, 4) exited non-zero
on 'SIP/dzalewski-081afaf0'
Above you are setting and clearing some database entries. What in your
dialplan are you using to act upon these values? You need something
resembling Example 1 on this page:
http://www.voip-info.org/wiki/view/Asterisk+call+forwarding Which
takes your saved values and acts on them.
Or perhaps I am misunderstanding something?
Steve
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Re: [asterisk-users] Zaptel Won't Compile In Slackware 11.0 Kernel 2.4.33
Hi, I observed that too. I already got that 2.6.x kernel and it is there actually. Though Patrick has put a 2.6.x kernel in /extra, I am still thinking if I will upgrade my existing kernel in this laptop. The maintainer of slackware did not made the 2.6.x as the default kernel for some other reasons he believes. I told you, the slackware 11.0 has a default kernel of 2.4.33.3. I am sticking with SVN because of development purposes. Does this mean that zaptel doesn't have backward compatibility with previous linux kernels? Is it a bug? Regards, Demuel Hi Demuel, Look, i think (im not very sure yet) that the *page-flags.h* file belongs to kernel = 2.5.x, not to the 2.4.x, Im using Slackware 10.2 , I have not upgraded yet to the 11.0 but i think that it comes with the 2.6.x kernel like a native kernel (not in /test/ directory anymore), 1. Why dont u try the 2.6.x kernel and get the kernel source ? 2. what's the result of runing this command on ur computer : *uname -a* 3. Have u try another version (not the SVN) ? [EMAIL PROTECTED] wrote: Hi, Even a default Slackware 11.0 with 2.4.33 kernel source failed to look for page-flags.h and I do not have a card that your are referring to because this is a development machine on a laptop. It used to work before but the current source tree which i get into a week ago started to break out for my case. Hi Demuel, 1st. Do you have a card that support 'zttranscode' ? if Yes, go ahead with this, if no, there's no use for you to be compiling it. 2nd. Do NOT do this: [EMAIL PROTECTED]:/usr/src/linux/include/linux$ sudo make updatedb Do this [EMAIL PROTECTED]:/usr/src/linux/include/linux$ updatedb (as root, or you will have to permit that command 'updatedb' in the sudoers list for the user 'demuel', in ur case) then: [EMAIL PROTECTED]:/usr/src/linux/include/linux$ slocate page-flags.h If u dont have it (slocate will certainly finds it if u do), then try to get it (of course, not just that file cause you could be missing another one in the farther process of compilation). Try to find out of what package or source where that file belongs to, and get it... J. Espinal [EMAIL PROTECTED] wrote: [EMAIL PROTECTED]:/usr/src/linux/include/linux$ pwd /usr/src/linux/include/linux [EMAIL PROTECTED]:/usr/src/linux/include/linux$ sudo make updatedb make: *** No rule to make target `updatedb'. Stop. [EMAIL PROTECTED]:/usr/src/linux/include/linux$ ls -la page-flags.h /bin/ls: page-flags.h: No such file or directory [EMAIL PROTECTED]:/usr/src/linux/include/linux$ Did i missed something down here? Weird thing is, even a fresh install of slackware produced the same kind of error. Actually, it used to be working about a week before I made a source upgrade. Any thoughts? Regards, Demuel On Tue, Feb 13, 2007 at 02:23:21PM +, J. Espinal wrote: make a 'updatedb' , and look for 'page-flags.h' , i think that you might be missing that file, under the include/ directory in the linux kernel source directory. J. Espinal, [EMAIL PROTECTED] wrote: Anybody, I have download asterisk 1.4 via svn. whem I compiled it, I got the following error: /lib/modules/2.4.33.3/build/include/asm/system.h:190: warning: dereferencing type-punned pointer will break strict-aliasing rules zttranscode.c:37:30: linux/page-flags.h: No such file or directory make[1]: *** [zttranscode.o] Error 1 make[1]: Leaving directory `/home/kingkong/code/projects/asterisk/source/zaptel-1.4' make: *** [all] Error 2 make a 'updatedb' , and look for 'page-flags.h' , i think that you might be missing that file, under the include/ directory in the linux kernel source directory. Better yet: simply don't build zttranscode, unless you have a card that actually supports it... -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Best FXO Gateway
On Feb 15, 2007, at 1:12 AM, jameson asterisk wrote: Can anyone provide a recommendation based on user experience? Feel free to suggest an alternative gateway if one stands out. I've been working with the Grandstream GXW-4108 (the 8 port version of the 4108), and it was a rough start, but I *think* all my issues have been worked out. Initial setup wasn't too bad, it helped that I found someone else's notes on it on the Trixbox forum. It it hadn't been for two problems, I'd have probably been up and running with it in just a few hours. The two issues I had were 1: I had major logging problems that I originally blamed on the GXW, but turned out to by my syslog server. When I changed to a different syslog server, the GXW's logs started working fantastically. I needed that logging to debug the 2nd issue. The 2nd problem was more involved, the GXW didn't like my PSTN connection. It worked wonderfully when connected to my VoIP ATA, but when I went to PSTN, it had all sorts of problems. I had some back and forth dialog with Grandstream and they think they found the problem and fixed it, and sent me a beta test firmware to try out. I put that online yesterday, and so far it has been working fine, but yesterday my office was closed due to snow, so I wasn't able to really stress test it (but I also was not able to reproduce my PSTN connection problem, which previously I could do with ease, so it gives me hope that the problem is indeed fixed). I'll know better today when the office is open and I expect the 8 lines to be in use for a good part of the day with a solid mix of inbound and outbound calling. If all goes well, then I might be able to recommend the GXW-410x as a viable unit. However, it does have one feature that might be a show stopper for some. It selects the next outbound FXO port in a round robin manner. There does not appear to be port level control over which FXO port is used on a given outbound call. This is probably fine for most of the targeted users (going on price, I'd say they are aiming at the SoHo market, which is likely to have a single bank of numbers in a single hunt group, so round robin would work fine). But for some, this could be a show stopper and prevent them from being able to use the unit. I personally have to see if it is going to work for me as I actually have two different hunt groups in my 8 lines, so round robin is less than ideal for me as it can cause one group to busy out from outbound calls, while the other has no calls at all. I do plan to send a feature request to Grandstream to give better control over selecting outbound ports. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk freeze due to too many open file error
Hi, my Asterisk 1.2.9.1 suddenly freezed (stop now did not work!!) . I found the following errors inside /var/log/asterisk/message: Feb 14 14:55:41 ERROR[11273] rtp.c: Unable to allocate socket: Too many open files Feb 14 14:55:41 WARNING[11273] chan_sip.c: Unable to create RTP audio session: Too many open files Feb 14 14:55:43 WARNING[11273] acl.c: Cannot create socket Feb 14 14:55:46 WARNING[11273] acl.c: Cannot create socket Feb 14 14:56:01 WARNING[11283] chan_misdn.c: Write returned =0 (err=Destination address required) Feb 14 14:59:56 WARNING[13725] res_agi.c: unable to create fromast pipe: Too many open files Feb 14 15:01:21 WARNING[13729] res_agi.c: unable to create fromast pipe: Too many open files Feb 14 15:01:56 WARNING[13744] res_agi.c: Unable to create toast pipe: Too many open files Feb 14 15:01:56 ERROR[13744] cdr_custom.c: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : Too many open files Feb 14 15:01:56 ERROR[13744] cdr_csv.c: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many open files Feb 14 15:02:18 WARNING[13747] res_agi.c: Unable to create toast pipe: Too many open files Feb 14 15:02:18 ERROR[13747] cdr_custom.c: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : Too many open files Feb 14 15:02:18 ERROR[13747] cdr_csv.c: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many open files Considered that the result of ulimit is ulimited, is there anybody who knows how to avoid these errors? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi-calendar Overlay Layers?
Is there any calendar client that can point at OX for calendar data, which client can display multiple calendars simultaneously as *overlapping layers* in the GUI? With UI to de/select calendars from view, one by one. That is, a single grid of days displayed, with the events in each day displayed in the same day's view list, as if the layers were all events in a single calendar. And is there a way to get the OX Web interface to do this? Or a place in the source code that can be recoded to do it? Thanks. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call forwarding
you just post only call forward activation part of dialplan,
but you must also make dialplan part, that reflect, how is set this
callforward mark,
ie. if callforward is set, dial that number, if not, dial peer...
Dominik Zalewski wrote:
Hi All,
I'm using asterisk 1.2.15 and call forwarding doesnt work for me.
from my extensions.conf:
; Unconditional Call Forward
exten = _*21*X.,1,NoCDR
exten = _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4})
exten = _*21*X.,3,Playback(vm-saved)
exten = _*21*X.,4,Hangup
exten = #21#,1,NoCDR
exten = #21#,2,DBdel(CFIM/${CALLERID(NUM)})
exten = #21#,3,Playback(auth-thankyou)
exten = #21#,4,Hangup
debug from asterisk CLI:
-- Executing NoCDR(SIP/dzalewski-081afaf0, ) in new stack
Feb 15 15:00:19 NOTICE[32307]: cdr.c:443 ast_cdr_free: CDR on
channel 'SIP/dzalewski-081afaf0' not posted
Feb 15 15:00:19 NOTICE[32307]: cdr.c:445 ast_cdr_free: CDR on
channel 'SIP/dzalewski-081afaf0' lacks end
-- Executing Set(SIP/dzalewski-081afaf0, DB(CFIM/200)=204) in new
stack
-- Executing Playback(SIP/dzalewski-081afaf0, vm-saved) in new stack
-- Playing 'vm-saved' (language 'en')
-- Executing Hangup(SIP/dzalewski-081afaf0, ) in new stack
== Spawn extension (from-internal, *21*204, 4) exited non-zero
on 'SIP/dzalewski-081afaf0'
Thank you in advance,
Dominik
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Re: [asterisk-users] asterisk freeze due to too many open file error
On Thu, Feb 15, 2007 at 02:57:59PM +0100, Giorgio Incantalupo wrote: Hi, my Asterisk 1.2.9.1 suddenly freezed (stop now did not work!!) Could you kill the asterisk process directly? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk freeze due to too many open file error
we have this problem. In our case it was due to the voice mail app; it was failing to unlink files in memory when creating mp3s. Not sure what your specific problem might be Giorgio Incantalupo wrote: Hi, my Asterisk 1.2.9.1 suddenly freezed (stop now did not work!!) . I found the following errors inside /var/log/asterisk/message: Feb 14 14:55:41 ERROR[11273] rtp.c: Unable to allocate socket: Too many open files Feb 14 14:55:41 WARNING[11273] chan_sip.c: Unable to create RTP audio session: Too many open files Feb 14 14:55:43 WARNING[11273] acl.c: Cannot create socket Feb 14 14:55:46 WARNING[11273] acl.c: Cannot create socket Feb 14 14:56:01 WARNING[11283] chan_misdn.c: Write returned =0 (err=Destination address required) Feb 14 14:59:56 WARNING[13725] res_agi.c: unable to create fromast pipe: Too many open files Feb 14 15:01:21 WARNING[13729] res_agi.c: unable to create fromast pipe: Too many open files Feb 14 15:01:56 WARNING[13744] res_agi.c: Unable to create toast pipe: Too many open files Feb 14 15:01:56 ERROR[13744] cdr_custom.c: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : Too many open files Feb 14 15:01:56 ERROR[13744] cdr_csv.c: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many open files Feb 14 15:02:18 WARNING[13747] res_agi.c: Unable to create toast pipe: Too many open files Feb 14 15:02:18 ERROR[13747] cdr_custom.c: Unable to re-open master file /var/log/asterisk/cdr-custom/Master.csv : Too many open files Feb 14 15:02:18 ERROR[13747] cdr_csv.c: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Too many open files Considered that the result of ulimit is ulimited, is there anybody who knows how to avoid these errors? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call forwarding
On Thursday 15 February 2007 04:00:52 pm Pavel Jezek wrote: you just post only call forward activation part of dialplan, but you must also make dialplan part, that reflect, how is set this callforward mark, ie. if callforward is set, dial that number, if not, dial peer... Do you have any example of this diaplan part? Thanks, Dominik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Feeding digit input to PauseQueueMember
Hello, I'm trying to figure out how to do something that I hope is pretty easy. I have a remote phone system (Definity ProLogix) connected to my Asterisk system via a T1 cable (all onsite). I'd like to get some of these users on a queue hosted on the Asterisk. I've got it setup so that it seems to work OK (calls flow normally), but I'd like the users to be able to dial one extension to run PauseQueueMember, and another to do UnpauseQueueMember. Is something like this possible? Answer Playback (what extension to pause) Get input --- how do I do that? PauseQueuemember (input from user) Playback (agent paused) Hangup I have done most of this already in other contexts, but I cant figure out how to get input from the user? Is there a function for that? What is it? Thanks, Stefano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP response 482 Loop Detected
Any news about this ? Mohamed Farid ,, Telecommunication Security Section Head ,, Mediterranean Smart Cards Company ,, 92 Tahreer Street. Dokki / Cairo / Egypt Website: www.mscc.com.eg http://www.mscc.com.eg/ Email : [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Phone : +2 02 3331439/+2 02 3331400 Fax : +2 02 7621164 Mobile : +2 0122258350 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mohamed Farid Sent: Wednesday, February 14, 2007 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP response 482 Loop Detected I have a Cisco Call Manager - and need to use the IVR Feature from Asterisk. My extension is 400 and I am calling 558 on Asterisk In my extension.conf I have these lines : exten = 558,1,Answer exten = 558,2,Playback(message.wav) exten = 558,3,Dial(SIP/[EMAIL PROTECTED]) When I call 558 I heared the message then Asterisk tries to call 439 on CallManager but with this error : -- Called [EMAIL PROTECTED] -- Got SIP response 482 Loop Detected back from CallManager -- Now forwarding SIP/CallManager-097b3dc0 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/CallManager-1781) == Everyone is busy/congested at this time (1:0/0/1) How can I overcome this ... Mohamed Farid ,,, This e-mail (including attachments) is classified as Mediterranean Smart Cards Company confidential and proprietary information The recipient hereby is committed to hold in strict confidence the contents of this (e-mail, document, and information) and not to disclose to any third party without the prior written consent of Mediterranean Smart Cards Company. Recipient will be held liable for any unauthorized disclosure. It is intended solely for the addressee. Unless you are the addressee, you may not read, copy, use or store this e-mail in any way, or permit others to. If you have received it in error, please notify the sender by return e-mail and delete the message in its entirety, including any attachments ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hint and CallerID
Hi, I use two hint-extensions to monitor my two ISDN-Lines: exten = 10,hint,Zap/10 exten = 11,hint,Zap/11 My Snom subscribed to the hints and then one line gets busy i have a LED assigned to the line, that flashes til the call is up and then stay on til the call is over. So far so good. If a call comes in, the snom displays also, a 10 or a 11. Is there any chance that i am able to display the callerid (if there is any) and (that would be great) the number he dialed ? Does the Subscription/Notify-System carry those information ? Thanks in advance ... Tobias ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - IP Network Call Recording
Apologies in advance as this is not directly Asterisk related, however I thought I might be able to leverage the experience of particiapants on this listserv for some advice. I have a client who is utilizing Talkswith PBX appliances, which have no native call monitoring/call recording capabilities. They are looking for an additional application, service or appliance that can sit on the LAN, and allow an administrator to monitor or recording inbound/outbound calls. If anyone is aware of a mechanism or solution that would provide this capability, please shoot me an email. Thanks Cory Andrews ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange behaviour with Dial cmd
Ok thank you a lot!!! On 2/15/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Il Neofita [EMAIL PROTECTED] Date: Thu, 15 Feb 2007 03:37:14 -0500 But I tought that hangup was suppose to close the call, however, is not the case and a really did not catch why. Now I see where the confusion comes from. Asterisk doesn't really speak English - or Chinese for that matter:-) In telephony, there is no way for the callee to tell the caller to stop ringing - unless you answer it first. Once you answer, you can do a number of things, the rudest being to immediately hang up. (I saw live people doing this intentionally.) Your only other option really is to ignore. I just thought up this simple method to ignore: divert the dial plan to simply Wait() an unreasonable amount of time in hope that the caller hangs up. exten = s,1,Dial(yourcell,5) exten = s,n,Wait(300) That's assuming your provider provides disconnect supervision. You can also Play(prerecorded,noanswer) if your provider supports it. (Won't hurt to try even if not - but disconnect supervision is a must.) Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Call Start
Brian Capouch wrote: Stephen Bosch wrote: And use a different Wiki engine! Augh! (Mediawiki, anyone?) Who runs voip-info.org? I'll bet if you volunteered to take it over, the folks who run it would gladly let you have it And I'd further bet they'd gladly let you run whichever Wiki software you want!! Otherwise, it strikes me as unseemly for you to criticize the way it's being done. Why do you think I asked the question? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - IP Network Call Recording
Check out CallRex, they list Talkswitch as a supported product (also Asterisk): http://www.telrex.com/callrex.htm I've seen it being used with Cisco phones on a hosted Covad environment and it is pretty neat. (I have no affiliation with them whatsoever). Cory Andrews wrote: Apologies in advance as this is not directly Asterisk related, however I thought I might be able to leverage the experience of particiapants on this listserv for some advice. I have a client who is utilizing Talkswith PBX appliances, which have no native call monitoring/call recording capabilities. They are looking for an additional application, service or appliance that can sit on the LAN, and allow an administrator to monitor or recording inbound/outbound calls. If anyone is aware of a mechanism or solution that would provide this capability, please shoot me an email. Thanks Cory Andrews ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Network stuff you didn't know http://www.networkoblivion.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel Won't Compile In Slackware 11.0 Kernel 2.4.33
Hi again, [EMAIL PROTECTED] wrote: Hi, I observed that too. I already got that 2.6.x kernel and it is there actually. Though Patrick has put a 2.6.x kernel in /extra, I am still thinking if I will upgrade my existing kernel in this laptop. The maintainer of slackware did not made the 2.6.x as the default kernel for some other reasons he believes. I told you, the slackware 11.0 has a default kernel of 2.4.33.3. I am sticking with SVN because of development purposes. Yes you are right there about the default kernel, Does this mean that zaptel doesn't have backward compatibility with previous linux kernels? Is it a bug? I'm starting to think that it could be a backward version compatibility problem too, I set up a server with Asterisk some days ago using 1.4 version, but it was with CentOS, and had the 2.6.x kernel by default. I think i should make some test at home about that issue (I use it at home), Tell me if u find something about that, Regards, Demuel take care, J. Espinal ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk freeze due to too many open file error
Hi Tzafrir, it was the only solution. I had to kill Asterisk and restart it. I've got many PBX installed but this is the first time it happened. I've searched for some opened file limit in linux but found nothing and ulimit says unlimited. Giorgio Incantalupo Tzafrir Cohen wrote: On Thu, Feb 15, 2007 at 02:57:59PM +0100, Giorgio Incantalupo wrote: Hi, my Asterisk 1.2.9.1 suddenly freezed (stop now did not work!!) Could you kill the asterisk process directly? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No Ringback, only on 1 SIP provider
Hi, I have the following situation: At a branch , there is a Cisco Call Manager with users all having Cisco phones. Now I put down a Asterisk 1.2.12 box at the branch, which talks H323 via chan_oh323 to the CCM. So calls go from the CCM, go H323 to the local Asterisk box, then I take it via SIP to another Asterisk box. From there I am hooked up to 2 different providers, for Local and International, both via SIP. The problem I am having is that the users dont get ringback (ringing indication) when they dial International numbers, yet it works perfectly when they dial Local numbers. Yet, to test, from a hardphone plugged into Asterisk2, I get ringback, so its not the Interntional provider, it must be the SIP trunk from Asterisk1 to Astrisk2. (ringback) NationalProvider | SIP| | H323 SIP | SIP (no ringback) Users phones - CCM 4.1 Asterisk1-Asterisk2-InternationalProvider | | ZAP hardphone Here is the sip.conf from Asterisk1. [N_G] type=friend host=10.255.255.1 username=N_G secret=N_G disallow=all allow=g729 canreinvite=no qualify=yes progressinband=yes (tried this yes/no/never, made no difference) When I call goes from Asterisk1 to Asterisk2, I get the 'making progress passing it to xxx', but I dont hear ringing, then the person answers. -- thanks, Yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-calendar Overlay Layers?
Sorry, I sent that message to the wrong list. Tho if you know the answer, please don't let that stop you from emailing it to me :). On Thu, 2007-02-15 at 08:21 -0700, [EMAIL PROTECTED] wrote: Date: Thu, 15 Feb 2007 08:54:43 -0500 From: Matthew Rubenstein [EMAIL PROTECTED] Subject: [asterisk-users] Multi-calendar Overlay Layers? To: Asterisk-Users asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain Is there any calendar client that can point at OX for calendar data, which client can display multiple calendars simultaneously as *overlapping layers* in the GUI? With UI to de/select calendars from view, one by one. That is, a single grid of days displayed, with the events in each day displayed in the same day's view list, as if the layers were all events in a single calendar. And is there a way to get the OX Web interface to do this? Or a place in the source code that can be recoded to do it? Thanks. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - IP Network Call Recording
if you can't use asterisk for recording ;-) you can try zoom-int callrec, this works listening on switch span port to record calls... but it's not free app Cory Andrews wrote: Apologies in advance as this is not directly Asterisk related, however I thought I might be able to leverage the experience of particiapants on this listserv for some advice. I have a client who is utilizing Talkswith PBX appliances, which have no native call monitoring/call recording capabilities. They are looking for an additional application, service or appliance that can sit on the LAN, and allow an administrator to monitor or recording inbound/outbound calls. If anyone is aware of a mechanism or solution that would provide this capability, please shoot me an email. Thanks Cory Andrews ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pause a Audio File Problem
Hello all .I had one question that, Is it possible to pause a audio file with out passing any escape digits. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interruptible announcements in queue application
Shouldn't you be putting your information in the music-on-hold, rather than the queue announcement? Matt wrote: I am not aware of one.Why would you want your queue announcement interupted? When we had our Nortel, I found that feature annoying because people would be transfered to the agent half way through a message. Confusing. I configured it to not break out of an annoucement. On 2/15/07, [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello all, I've found another issue with the queue application. Assuming I've configured a queue with a long periodic announcement and have two queue members assigned. Both queue members are busy at a time, while another caller is joining the queue. After a while the periodic announcement is played back to the caller, in that case it takes about 40 seconds to be played back. If then one of the two agents becomes available, the call is unfortunately not routed to the agent, until the playback of the announcement has finished. If you display the agent status and the queue to a supervisor he can see that there are callers waiting up to 40 seconds, even if there are available queue members. For inbound call centers tat is more than suboptimal. Does anybody know if somebody already created a patch to interrupt queue announcement when an agent becomes ready? Thanks and Regards Markus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] End Wrap-up Time?
We also have not managed to find a solution. Personally, I dunno why the agents want to stop wrap. I could see what administratively you might want them to. But for some reason our agents actually wanted to.Anyway, I created a button that says Wrap Cancel. It does nothing but play a sound file. They're all happy :) LOL.Anyway, it would be nice to have this feature on future verison of Asterisk. On 2/15/07, Rob Hillis [EMAIL PROTECTED] wrote: Hi James, The only solution I've managed to find so far is to set the wrap-up time to 5 seconds and tell the operators that if they need more time, they need to put themselves on pause. See PauseQueueMember and UnpauseQueueMember. If someone has a better solution, I'd be most pleased to hear of it! James Fromm wrote: Does anyone have a solution to allow an agent to selectively end his wrap-up time? We define a wrap-up time of 60 seconds to allow our agents to finish their notes from a call. In some cases, the full 60 seconds is not needed and our agents would like to end their wrap-up time. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Maximum Number of Calls Asterisk Can Handle
Can anyone share their experience on the maximum number of calls a given asterisk box/asterisk software can handle? I see the asterisk business edition can handle up to 240 simultaneously with appropriate licensing, but that doesn't seem to be many at all. For now, I plan to use the stable open source versions - would it be reasonable to say that it is more of hardware limitation on the number of calls that can be made simultaneously, or are there only so many calls the asterisk software programming is equipped to handle simultaneously? Also is the asterisk software written to effectively take advantage of multiple processors? The systems I plan to use for asterisk have the following specs: dual 2.8GHZ+ Pentium's 2GB RAM+ Gigabit interfaces In my situation, I have no plans to run anything other than G.711/SIP, so my transcoding need is probably only limited to the playback of pre-recorded messages as well as any processing involved in leaving voicemail. in theory, a gigabit interface can move 1048576Kbit/sec - now if i generously allocate 96Kbit/sec for every G.711 call, the network transport can handle, again in theory, 10922 simultaneous calls. would it be wrong to expect performance near this mark for the asterisk software? Feedback appreciated, -- --- John C. Wolosuk Jr. Unix/Linux Systems Administrator Academic Computing Communications Center University of Illinois @ Chicago E-Mail: jwolosuk at uic dot edu --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] addons 1.4 and cdr_addon_mysql not installed !
So, after reading this, I wonder if anyone has 1.4 and MySQL working... Is there a non-standard version I can download? more /usr/src/asterisk-1.4.0/doc/mysql.txt MYSQL LICENSING UPDATE == We were recently contacted by MySQL and informed that the MySQL client libraries are now under GPL license and not LGPL license as before. Since Asterisk does allow exceptions to GPL, we are removing MySQL support from standard Asterisk. We will, where appropriate, make it available via a separate package which will only be usable when Asterisk is used completely within GPL (i.e. not in conjunction with G.729, OpenH.323, etc). We apologize for the confusion. You may find this in the new asterisk-addons package. Mark Spencer Digium -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Friday, January 05, 2007 12:13 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] addons 1.4 and cdr_addon_mysql not installed ! On Fri, Jan 05, 2007 at 05:44:28PM +0100, Luca Lafranchi Lists wrote: Hi, I have installed asterisk on Ubuntu 6.06 server CD All required packages has been installed and upgraded When start sudo make menuselect As a rule, make as a user, make install as root. No need for sudo for anything other than 'make install' and such. from addons, I can't select all addons that require mysqlclient (app_addon_sql_mysql, cdr_addon_mysql, res_config_mysql). If I run apt-cache search mysqlclient, I find the following installed packages: libmysqlclient15-dev - mysql database development files libmysqlclient15off - mysql database client library You need the -dev one installed (recall that you're building a package. The relevant build dependencies according to the current Etch package: libmysqlclient15-dev asterisk-dev (It also requires libsqlite3-dev, but res_sqlite3 has a broken build process anyway and cannot use the system version of sqlite3) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] End Wrap-up Time?
On Tuesday 13 February 2007 11:30 am, James Fromm wrote: Does anyone have a solution to allow an agent to selectively end his wrap-up time? We define a wrap-up time of 60 seconds to allow our agents to finish their notes from a call. In some cases, the full 60 seconds is not needed and our agents would like to end their wrap-up time. This is coming right out of left field, as I've never set up an Asterisk queue or agent system, but is it possible to pause and unpause while in the wrap-up time? What happens? Does the wrapup time go away then? Might be a counter-intuitive way around it if so... -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum Number of Calls Asterisk Can Handle
John C. Wolosuk Jr. wrote: in theory, a gigabit interface can move 1048576Kbit/sec - now if i generously allocate 96Kbit/sec for every G.711 call, the network transport can handle, again in theory, 10922 simultaneous calls. would it be wrong to expect performance near this mark for the asterisk software? Feedback appreciated, Yes, it would be wrong to expect performance near that mark. Most systems cannot handle the TCP processing load generated by a gigabit ethernet interface, let alone process everything that goes along with calls associated with that traffic. A TCP offloader engine will help, but the limitation is still within Asterisk itself. There is a lot that goes into processing everything related to a call. Now, if you can get the media to be re-invited to a media gateway, then you can handle significantly more calls. My experience is that when you are running media through an Asterisk server, 240 calls is an average maximum for a typical server. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum Number of Calls Asterisk Can Handle
in theory, a gigabit interface can move 1048576Kbit/sec - now if i generously allocate 96Kbit/sec for every G.711 call, the network transport can handle, again in theory, 10922 simultaneous calls. would it be wrong to expect performance near this mark for the asterisk software? 10922 on any currently available PC architecture? Nope. It's closer to 160 kpbs per call (two legs, 80 kbps each) in either direction. With 20 ms packet size, for 10922 calls you'd be looking at 2184400 packets/sec processed by Asterisk... I don't think so. Plus with 10922 calls and an average of 2 mins/call, you're looking at about 90 call setups/tear downs a second. I don't think even without running the RTP through Asterisk this box could handle 10922 concurrent calls. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk freeze due to too many open file error
On Thu, 15 Feb 2007, Giorgio Incantalupo wrote: Hi Tzafrir, it was the only solution. I had to kill Asterisk and restart it. I've got many PBX installed but this is the first time it happened. I've searched for some opened file limit in linux but found nothing and ulimit says unlimited. Unlimited is unlimited for the process currently running - ie. the process that you type the 'ulimit' command into. If you start asterisk as a user different from the one you are typing the commands into (eg. is it started as the user 'asterisk' rather than root?) then the limits _could_ be different, depending on how your system deals with this sort of thing. (PAM, or other subsystems) And in any case, there is an upper system limit set by the kernel. Try this: cat /proc/sys/fs/file-max This is set at boot time and is generally dependant on how much memory you have. You can change this number if required, but it usually isn't. You may also want to look at /proc/sys/fs/file-nr - the first number is the number of open files in the whole system. If the system is running away then I'd suggest looking deeper into it - is it opening a file and never closing it again, etc. Hard to track down unless you have a good knowlege of what's running, etc. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum Number of Calls Asterisk Can Handle
John C. Wolosuk Jr. wrote: Can anyone share their experience on the maximum number of calls a given asterisk box/asterisk software can handle? I see the asterisk business edition can handle up to 240 simultaneously with appropriate licensing, but that doesn't seem to be many at all. For now, I plan to use the stable open source versions - would it be reasonable to say that it is more of hardware limitation on the number of calls that can be made simultaneously, or are there only so many calls the asterisk software programming is equipped to handle simultaneously? Hardware plays a part as does what you are doing. The execution of different things for different tasks can also yield different scaling issues. First example: Simple channel comes in channel goes out with no media scenario. Asterisk is obviously going to scale better since it doesn't have to do as much. It's basically handling signalling and call setup/teardown. Second example: Simple channel comes in channel goes out with media. Media is going to be moved from kernel space to user space, sent to the other channel in an Asterisk frame (which may or may not allocate memory, depending on if you are using 1.4 and caching), turned into a proper frame to be sent out to network (in the case of RTP it gets RTP headers attached), moved to kernel space, and sent. This can be a very intensive process and doesn't scale as well as above obviously. Also is the asterisk software written to effectively take advantage of multiple processors? Yes, it is multithreaded and can take advantage of multiple processors. The systems I plan to use for asterisk have the following specs: dual 2.8GHZ+ Pentium's 2GB RAM+ Gigabit interfaces In my situation, I have no plans to run anything other than G.711/SIP, so my transcoding need is probably only limited to the playback of pre-recorded messages as well as any processing involved in leaving voicemail. Your issue is probably going to be hard disk access. For example: With minimal tweaking on my development machine I can get 330 channels up with full RTP in both directions playing back audio from a hard disk. If I move to a ramdisk based solution this goes up to 550. That's a 220 channel increase. Pushing the channels past this yields degraded audio quality. (AMD Athlon64 X2 4200+ with 1GB of RAM, 80GB SATA hard drive for those who are curious). The easiest way to know though is to setup your system and test it using something like sipp. Identify where your bottlenecks are and see what you can do to alleviate them. In the above scenario hard disk access was a bottleneck so I took it out of the picture and look what happened. in theory, a gigabit interface can move 1048576Kbit/sec - now if i generously allocate 96Kbit/sec for every G.711 call, the network transport can handle, again in theory, 10922 simultaneous calls. would it be wrong to expect performance near this mark for the asterisk software? In a perfect world maybe that would happen but this is a simple PBX running on Linux. Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connect PBX CO Port to TDM FXS Port
I would like to connect a Legacy PBX (Avaya IP Office 406) to an Asterisk Server. The Avaya has 3 CO Ports available. I thought buying a TDM30B card with 3 FXS ports to connect the * to the Avaya CO Ports. Is this the right approach? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions Daniel Kocher ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Fxo] Digium TDM01B vs. OpenVox A400P01?
On Thu, 2007-02-15 at 03:13 +0100, Vincent Delporte wrote: Hello If someone had the opportunity of trying those two analog cards, how do they compare? Digium's sells for $150 while OpenVox's sells for $95. OpenVox makes cheap knockoffs but they are virtually identical to the original. It uses the same zaptel module and Asterisk does not know the difference. They can even use the digium modules and viceversa! -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mini-ITX board + FXO PCI card?
Hello, Am Donnerstag, den 15.02.2007, 10:55 +0800 schrieb Leo Ann Boon: 1. The smallest mini-ITX case I found that accepts a PCI card is the Travla C138: If you used a mini-ITX with a Digium TDM400P, do you know if it fits? I didn't find its width, and apparently, the C138 will not accept a PCI card bigger than 17,52cm. The C137 can fit 2 TDM400P with the right riser. If You are using the riser card, there will be shared interrupts. The two slots of the riser card are using the same IRQ AFAIK. Karsten ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hint and CallerID
On Thu, 2007-02-15 at 16:11 +0100, Tobias Wolf wrote: Hi, I use two hint-extensions to monitor my two ISDN-Lines: exten = 10,hint,Zap/10 exten = 11,hint,Zap/11 My Snom subscribed to the hints and then one line gets busy i have a LED assigned to the line, that flashes til the call is up and then stay on til the call is over. So far so good. If a call comes in, the snom displays also, a 10 or a 11. Is there any chance that i am able to display the callerid (if there is any) and (that would be great) the number he dialed ? Does the Subscription/Notify-System carry those information ? Callerid is not defined by the hints. You need the line: callerid=asreceived This should be in the definition of your zap channel so it passes the callerid information without modification to your phones. -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk freeze due to too many open file error
If the system is running away then I'd suggest looking deeper into it - is it opening a file and never closing it again, etc. Hard to track down unless you have a good knowlege of what's running, etc. If you think it might be asterisk itself, then check which files it has open. lsof -p `ps h -C asterisk -o pid | head -1` -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk freeze due to too many open file error
If the system is running away then I'd suggest looking deeper into it - is it opening a file and never closing it again, etc. Hard to track down unless you have a good knowlege of what's running, etc. lsof -p `ps h -C asterisk -o pid | head -1` | grep -Fc '/dev/zap/timer' 120 You have to open your own timer device over one hundred times in the same process? -HJC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect PBX CO Port to TDM FXS Port
Daniel Kocher wrote: I would like to connect a Legacy PBX (Avaya IP Office 406) to an Asterisk Server. I would hardly consider the IP office a legacy PBX Unless, that is, you consider anything other than Asterisk legacy IP office is current competition for Asterisk, as is Call Manager You really need to define WHAT your goal is here. Provide a Voicemail for your IP office? or what?? John Novack The Avaya has 3 CO Ports available. I thought buying a TDM30B card with 3 FXS ports to connect the * to the Avaya CO Ports. Is this the right approach? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions Daniel Kocher ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues do not accept calls if all agent are busy?
On 2/15/07, Angel Heart [EMAIL PROTECTED] wrote: cud any one help me figuring out the problem... When the agent in a queue is engaged, it cannot accept anymore calls, below is the script; Angel, Check your queues.conf, specifically the joinempty parameter. See below the relevant part in the queues.conf sample file: ... ; This setting controls whether callers can join a queue with no members. There ; are three choices: ; ; yes- callers can join a queue with no members or only unavailable members ; no - callers cannot join a queue with no members ; strict - callers cannot join a queue with no members or only unavailable ; members ; ; joinempty = yes ... Cheers, -- Ex Vito ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk freeze due to too many open file error
On Thu, Feb 15, 2007 at 04:47:56PM +0100, Giorgio Incantalupo wrote: Hi Tzafrir, it was the only solution. I had to kill Asterisk and restart it. I've got many PBX installed but this is the first time it happened. I've searched for some opened file limit in linux but found nothing and ulimit says unlimited. The asterisk process did could not get an extra file descriptor. This may be because you have some ~1000 channels (or maybe less, if you use something based on h323?). An extra file descriptor is also needed to answer a manger connection and also needed to answer an rasterisk (asterisk -r) connection. Thus you need to kill the asterisk process directly. Anyway, killing a process with SIGTERM is not that bad. It does give the process the time to tidy up. Now, if you weren't uing so many channels, it is good to know whay file descriptors leaked. To do that, have a look occasionally at /proc/PID_OF_ASTERISK/fd . that directory shows all the current file descriptors the Asterisk process uses. Before killing that asterisk process, run: ls -l /proc/PID_OF_ASTERISK/fd fds_of_asterisk -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connect PBX CO Port to TDM FXS Port
I would like to connect a Legacy PBX (Avaya IP Office 406) to an Asterisk Server. The Avaya has 3 CO Ports available. I thought buying a TDM30B card with 3 FXS ports to connect the * to the Avaya CO Ports. Is this the right approach? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions Daniel Kocher ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - IP Network Call Recording
From: Pavel Jezek [EMAIL PROTECTED] Date: Thu, 15 Feb 2007 16:51:24 +0100 if you can't use asterisk for recording ;-) Cory didn't say that:-) Theoretically you can set up Asterisk in between Talkswitch and end points, map Talkswitch agents with Asterisk agents, then use Asterisk to monitor/record. Kinda clumsy and possibly costly, but should work as a basic application. It may not give you all the native monitoring bells and whistles, though. Talking off my hat. Yuan Liu you can try zoom-int callrec, this works listening on switch span port to record calls... but it's not free app Cory Andrews wrote: Apologies in advance as this is not directly Asterisk related, however I thought I might be able to leverage the experience of particiapants on this listserv for some advice. I have a client who is utilizing Talkswith PBX appliances, which have no native call monitoring/call recording capabilities. They are looking for an additional application, service or appliance that can sit on the LAN, and allow an administrator to monitor or recording inbound/outbound calls. If anyone is aware of a mechanism or solution that would provide this capability, please shoot me an email. Thanks Cory Andrews ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect PBX CO Port to TDM FXS Port
Daniel Kocher wrote: I would like to connect a Legacy PBX (Avaya IP Office 406) to an Asterisk Server. The Avaya has 3 CO Ports available. I thought buying a TDM30B card with 3 FXS ports to connect the * to the Avaya CO Ports. Is this the right approach? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions Daniel Kocher It kind of depends on what you're trying to accomplish. What do you want to be able to do with this connection? -Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Fxo] Digium TDM01B vs. OpenVox A400P01?
Carlos Chavez wrote: On Thu, 2007-02-15 at 03:13 +0100, Vincent Delporte wrote: Hello If someone had the opportunity of trying those two analog cards, how do they compare? Digium's sells for $150 while OpenVox's sells for $95. OpenVox makes cheap knockoffs but they are virtually identical to the original. It uses the same zaptel module and Asterisk does not know the difference. They can even use the digium modules and viceversa! Then they must suffer from the same issues with the PCI bus Probably a good choice for someone who wants to play around, and has a stack of old motherboards to try until one works. For a more trouble free installation AND a 5 year warranty, and real support, the Sangoma A200 wins out every time. JMO John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 7912 phones loosing registration
I have a handful of 7912's connected to my asterisk 1.2.14 server. (6 to be exact). I get the X on the display sometimes for loosing registration. I have the config file for the 7912's SipRegInterval: 60 and asterisk is the default. ; maxexpirey=3600 ;defaultexpirey=120 I've not changed them. How can I keep these phones online and stop loosing registration? Thanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connect PBX CO Port to TDM FXS Port
I would like to connect a Legacy PBX (Avaya IP Office 406) to an Asterisk Server. The Avaya has 3 CO Ports available. I thought buying a TDM30B card with 3 FXS ports to connect the * to the Avaya CO Ports. Is this the right approach? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions Daniel Kocher ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Feeding digit input to PauseQueueMember
Is something like this possible? Answer Playback (what extension to pause) Get input --- how do I do that? PauseQueueMember (input from user) Playback (agent paused) Hangup Eventually I found it: The Read Application http://www.asteriskguru.com/tutorials/read.html Or http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Read Stefano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] h323 - SIP conversion
I'm looking at setting up an asterisk box dedicated to SIP-H323 conversion (a 3rd party is currently converting the protocols for us). 1. Is it worthwhile to split this functionality onto a second server? Or should we let the ast pbx handle the conversion? (we have a couple hundred active channels to convert) 2. Is it better to go direct from SIP to AIX? 2. Can Asterisk handle H323 natively with problem? Thanks, MD ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
On Feb 15, 2007, at 3:17 AM, Wireless wrote: - Original Message - From: Nic Bellamy [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 13, 2007 8:40 PM Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC) Larry Shields wrote: I recently read about the following new technologies from Digium. Has anyone tried the new HPEC or knows when it will be available? It's out now, and I've tried it - the difference between HPEC and MG2 from trunk is stunning - in situations with bad echo where MG2 can take ten or more seconds to converge to a reasonable degree, HPEC does it in perhaps 300ms - converging on my intake of breath before I say hello, and absolutely no echo after that unless I purposefully go out of my way to screw it up (whistling/blowing into the handpiece for instance - even then, the malfunction is minimal). You can now buy it from the Digium website (US$10 per channel), or if you have an in-warranty Digium card, email through the serial numbers to Digium support and they'll give you a key (this is what I did). You'll need Zaptel 1.2.13 to make it go. It does take quite a bit of CPU though - perhaps 70% more compared to MG2-trunk for the same number of taps from my rough measurements. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Does anyone know if the HPEC will work on a Sangoma A200 / 2 port FXO card? (I'm assuming so as it still uses Zapel) I've 2 PSTN lines one of which I cannot get rid of the echo, I've tried a 2GHz machine as apposed to my normal P3 650MHz and this made no difference. Would the 650Mhz be enough to run HPEC on one line (I assume only needing one licence) This is what Digium say on their web site: Digium recommends that users requiring 8 channels at 1024 taps run a PC comparible to a 3.0 GHz Pentium 4, while users only requiring 4 channels at 1024 taps may run a 2.5 GHz Pentium Celeron. The CPU requirements are such that it is impractical to operate this echo canceller at 1024 taps for a full T1 or E1 of channels. Yes, it will work with any card that uses zaptel. You just have to pay a per port fee to use it with a non-digium card. I'm not sure about the performance requirements for one port though. You could try it with a low tap count and keep bumping up your taps (echocancel=[32,64,128,256,512,1024]) until it cancels the echo though. That way you'd only use as much cpu as you absolutely have to. Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk guru wanted, SoCal (LA/OC/San Bernardino County)
We've mostly gotten our Asterisk install working, but there are a couple glitches I haven't been able to fix. I'm looking for someone who knows Asterisk, can do some consulting work, and is in Southern California. Los Angeles or Orange County are ok, but I'd prefer someone in the Inland Empire, with strongest preference given to people here in the High Desert or in the San Bernardino/Fontana/Rialto/ Rancho Cucamonga area. Email me off-list: [EMAIL PROTECTED] Thanks -- Steve Sobol, Professional Geek ** Java/VB/VC/PHP/Perl ** Linux/*BSD/Windows Victorville, California PGP:0xE3AE35ED It's all fun and games until someone starts a bonfire in the living room. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - IP Network Call Recording
On 2/15/07, Cory Andrews [EMAIL PROTECTED] wrote: Apologies in advance as this is not directly Asterisk related, however I thought I might be able to leverage the experience of particiapants on this listserv for some advice. I have a client who is utilizing Talkswith PBX appliances, which have no native call monitoring/call recording capabilities. They are looking for an additional application, service or appliance that can sit on the LAN, and allow an administrator to monitor or recording inbound/outbound calls. If anyone is aware of a mechanism or solution that would provide this capability, please shoot me an email. Thanks Cory Andrews Cory, From their website it appears they are using SIP. With any luck it will be SIP + ulaw (without re-invites). If so, do this: 1) Get a decent managed switch that can setup monitor ports. Configure one port to monitor the port connected to the Talkswitch. 2) Get a decent dual-homed machine. 3) Connect one interface of the dual-homed machine to the monitor port. Running Linux, do an ifconfig up [interface name] (no IP address). Configure the other interface to connect to a network for management, copying files, etc. 4) Start up tcpdump on the interface, writing to a file. 5) Use something like Cain + Abel to read the RTP and dump the audio to a file. 6) Convert files to desired format using sox. The only step I left out was Profit!. Seriously though, this depends on a few key assumptions about the Talkswitch: 1) That it is standard SIP. 2) It uses ulaw. 3) It doesn't do re-invites. Not any one of these is a show stopper for this type of sollution, but any one of them (or all of them) could make life a bit harder for you... Good luck! -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Long call setup times on SIP to zaptel calls
I have had a lot of complaints about the time it takes to setup a call. I have timed it and it is almost five seconds before it even starts ringing. The SIP device sends the request almost instantly but the channel is taking a long time to pickup and dial. It wouldn't be so bad but they hear nothing. I would like to provide ringback before the zaptel actually starts ringing the channel. Has anybody done this, it seems like it would be a zaptel option. Jordan Novak ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Long call setup times on SIP to zaptel calls
From: Jordan Novak [EMAIL PROTECTED] Date: Thu, 15 Feb 2007 13:45:39 -0600 I have had a lot of complaints about the time it takes to setup a call. I have timed it and it is almost five seconds before it even starts ringing. The SIP device sends the request almost instantly but the channel is taking a long time to pickup and dial. It wouldn't be so bad but they hear nothing. I would like to provide ringback before the zaptel actually starts ringing the channel. Has anybody done this, it seems like it would be a zaptel option. Jordan Novak A local ring back is built in Dial()'s r option. The problem with this is that the caller is never going to hear the real ring back. I haven't figured out a good strategy to deal with this caveat. Suggestions? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Long call setup times on SIP to zaptel calls
do your sip phones dial after a timeout? If the timeout is set to around 5 seconds you may have a dialplan issue on your sip hardware. Shane On 2/15/07, Jordan Novak [EMAIL PROTECTED] wrote: I have had a lot of complaints about the time it takes to setup a call. I have timed it and it is almost five seconds before it even starts ringing. The SIP device sends the request almost instantly but the channel is taking a long time to pickup and dial. It wouldn't be so bad but they hear nothing. I would like to provide ringback before the zaptel actually starts ringing the channel. Has anybody done this, it seems like it would be a zaptel option. Jordan Novak ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Long call setup times on SIP to zaptel calls
I only say this because nobody in our office knew how to use the checkmark on snom phones to initiate a call, they always just waited for the phone to initiate the call for them :) On 2/15/07, Shane Spencer [EMAIL PROTECTED] wrote: do your sip phones dial after a timeout? If the timeout is set to around 5 seconds you may have a dialplan issue on your sip hardware. Shane On 2/15/07, Jordan Novak [EMAIL PROTECTED] wrote: I have had a lot of complaints about the time it takes to setup a call. I have timed it and it is almost five seconds before it even starts ringing. The SIP device sends the request almost instantly but the channel is taking a long time to pickup and dial. It wouldn't be so bad but they hear nothing. I would like to provide ringback before the zaptel actually starts ringing the channel. Has anybody done this, it seems like it would be a zaptel option. Jordan Novak ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best phone for easy provisioning
We use Linksys/Sipura phones, and do mass provisioning via tftp and http. There is no need for a compiler for the SPA-841, 941, 942, 3000, or 2000 phones at least; I don't have direct experience with others. We feed a raw XML configuration file to the phone via a cgi-bin script which receives the MAC address as a form parameter, and all is well with the world. I posted our experiences on voip-info.org, here: http://www.voip-info.org/wiki/view/sipura+mass+deployment We've had our deployment system in place almost totally unchanged for the last 18 months or so with no real problems. The only thing I find slightly less than optimal is that for major configuration changes, the phones seem to need a factory reset to pick up the changes in a timely manner. Alan Ferrency On Mon, 12 Feb 2007, George Pajari wrote: Aastra are a delight -- no need for a compiler (like the Grandstream and Linksys phones) -- and extremely well documented configuration files. -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect PBX CO Port to TDM FXS Port
Daniel Kocher wrote: I would like to connect a Legacy PBX (Avaya IP Office 406) to an Asterisk Server. The Avaya has 3 CO Ports available. I thought buying a TDM30B card with 3 FXS ports to connect the * to the Avaya CO Ports. Is this the right approach? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions Daniel Kocher It kind of depends on what you're trying to accomplish. What do you want to be able to do with this connection? -Dave I would like to use the * as VoIP Gateway. Something like that: A user takes off a phone on a Avaya extension and dials for example 8 to reach the CO Port. Then Asterisk answers and sends a dial tone. The user dails a numer and Asterisk is doing the rest! (Sending the call to an SIP or IAX Provider) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
Andrew Kohlsmith wrote: On Thursday 15 February 2007 6:51 am, Steve Underwood wrote: It looks like octasic have started supplying their echo canceller as host software for zaptel now. I expect either canceller would work with the Sangoma cards, as they currently sit in the zaptel framework too. Out of curiosity, why do you suppose that it is the Octasic algorithm which is used in Digium's HPEC? I have no reasons to suspect otherwise, but I'm curious as to your reasons for suspecting that is indeed the case. I think Steve meant Octasic are _also_ now supplying their EC as host software for Zaptel. The HPEC canceller is from Adaptive Digital. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
Wireless wrote: Does anyone know if the HPEC will work on a Sangoma A200 / 2 port FXO card? (I'm assuming so as it still uses Zapel) I've 2 PSTN lines one of which I cannot get rid of the echo, I've tried a 2GHz machine as apposed to my normal P3 650MHz and this made no difference. Would the 650Mhz be enough to run HPEC on one line (I assume only needing one licence) It should work, providing all the Wanpipe stuff is ready to work with Zaptel 1.2.13. As far as performance, you should be able to get one, maybe two channels of 1024 tap cancellation on the P3, but I'd advise careful testing, perhaps even using oprofile for a while to keep an eye on what's using what. You also have to watch out extra carefully due to the following: HPEC works in sparse mode, meaning it can cover 1024 taps, but just cancels echo in the parts where there is echo - hence CPU usage will likely change quite a bit with different echo paths - ie. a simple single reflection path will use less CPU than a complicated path with more than one reflection. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Long call setup times on SIP to zaptel calls
Jordan said the SIP device sends the request almost instantly so it's not the SIP phone's fault. The channel bank probably takes 1-2 seconds to pick up and wait/check for dial tone, 1-2 second dialing, and the telco takes 1-2 second to ring. So the complete PDD is ~5 seconds. You could try putting a Ringing(); before the dial statement to let the SIP phone know the call is being connected. I believe once progress comes from the Dial command, it will replace the Ringing. However, if your channel bank answers the call right away, this won't help. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AgentCallBackLogin vs AddQueueMember
Hello, In our investigation of the AddQueueMember vs. AgentCallbackLogin situation, the major loss with using the published AddQueueMember replacement is that it assumes each agent is always using the same phone. We were not implementing agents this way at all. In fact the _only_ thing we really needed the agent code to do is allow a person to log into any one of multiple phones, and retain their agent extension number independent of the phone they're using. When they aren't logged in, they can't receive calls; when they are, they're always reached at the same extension. In short: I don't want a one to one mapping from extension numbers to telephones. I want a one to one mapping between extension numbers and _people_. If I had that, I would not need the AddQueueMember behavior: I simply keep all agents in all queues as before, and when they aren't logged in, they don't receive calls. Basically, the AddQueueMember solution provided in the Asterisk 1.4 documentation solves a different problem than many AgentCallbackLogin users were solving with their use of the Agent channel. I don't know exactly why AgentCallbackLogin is being deprecated, but in my experience, sufficiently new versions of Asterisk have serious deadlock bugs when using AgentCallbackLogin with other specific functions. This means we are stuck in the dark ages of Asterisk 1.2.3, because this is the last version we know of that fills our needs without deadlocking and causing unnecessary downtime. I hope this helps, Alan Ferrency On Wed, 14 Feb 2007, gc wrote: So you have to hard code the each queue name in the dialplan for an agent to login. What about hundreds of agents login 30-40 different queues? If this is the only way to do it, I will not use AddQueueMember at all. I do not know the reason for deprecating AgentCallBackLogin. But I do think remove it without appropriate replacement is bad idea. Gary ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect PBX CO Port to TDM FXS Port
I would like to use the * as VoIP Gateway. Something like that: A user takes off a phone on a Avaya extension and dials for example 8 to reach the CO Port. Then Asterisk answers and sends a dial tone. The user dails a numer and Asterisk is doing the rest! (Sending the call to an SIP or IAX Provider) John Novack [EMAIL PROTECTED] wrote: Daniel Kocher wrote: I would like to connect a Legacy PBX (Avaya IP Office 406) to an Asterisk Server. I would hardly consider the IP office a legacy PBX Unless, that is, you consider anything other than Asterisk legacy IP office is current competition for Asterisk, as is Call Manager You really need to define WHAT your goal is here. Provide a Voicemail for your IP office? or what?? John Novack The Avaya has 3 CO Ports available. I thought buying a TDM30B card with 3 FXS ports to connect the * to the Avaya CO Ports. Is this the right approach? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions Daniel Kocher - Sucker-punch spam with award-winning protection. Try the free Yahoo! Mail Beta.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Queues Problem
Help! I'm (still) having issues with Asterisk Queues. I want to implement a queue so that callers get the 'all our staff are busy at the moment, your call has been placed in a queue and will be answered by the first available operator. You may press 1 at any time to leave a voicemail' announcement, then they can press 1 and leave a voicemail. Documentation with * 1.2.x (I'm using 1.2.15) and in the O'Reilly Asterisk book says I can add a line context=blah to the queue definition and this becomes the 'escape context' where pressing buttons will take you to whilst in the queue. I've done this, and put the relevant context in extensions.conf and put extension 1 in there - and nothing happens - I call into the queue and press 1 and don't go anywhere. Please help if you know how to solve this issue, I have been working on it for a week and it's becoming quite urgent (not to mention causing me to tear my hair out with frustration...) Regards, John Breen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Guest registration in SIP
I remember seeing some way to allow unknown clients to register in Asterisk, but can no longer find any reference to such. Pointers? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Native format prompts
Hi all, I am trying to implement native format (ulaw) voice prompts and music on hold. Different documentation has different file extensions. Does Asterisk recognise them all? So far I have .ulaw .ul .pcm . Which should I use so Asterisk recognises them as native uLaw files ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Debugging a SIP / AudioCodes Problem
Audiocodes blatently violates the GPL... dont use their gear. On 11 Feb 2007 19:11:51 -, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I have 2 identical AudioCodes MP-112s. They have the same config except for the SIP usernames/passwords and the device IP. The configs in extension.conf and sip.conf are also identical. On one box, when I pick up the phone, I get a fast busy and the logs/debug show an automatic hangup. On the other device, I can make calls without a problem. I can even call the phone that can't make a call. Any ideas where I could start to figure out where the problem is? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] End Wrap-up Time?
I tried that. It didn't work :( On 2/15/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 13 February 2007 11:30 am, James Fromm wrote: Does anyone have a solution to allow an agent to selectively end his wrap-up time? We define a wrap-up time of 60 seconds to allow our agents to finish their notes from a call. In some cases, the full 60 seconds is not needed and our agents would like to end their wrap-up time. This is coming right out of left field, as I've never set up an Asterisk queue or agent system, but is it possible to pause and unpause while in the wrap-up time? What happens? Does the wrapup time go away then? Might be a counter-intuitive way around it if so... -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme - is this statement from the Wiki still true?
The conference bridge runs Ulaw codec by default. If you let people connect with GSM or other codecs, Asterisk will use CPU power to convert audio between codecs ... What about alaw channels is there any transcoding work being done there? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] colors in the console
You seem to start asterisk with safe_asterisk. That script
starts asterisk on a console of its own. Maybe it wa done to
allow the use of colors.
If you want a plain 'asterisk' to run with colors, try the patch in
http://bugs.digium.com/view.php?id=9048
Hi, Tzafrir ... The patch you submitted doesn't work with earlier
1.2 versions of Ast due to renaming of variables... This version does work
in those cases:
/*--- begin -*/
--- term.c 2006-12-13 09:24:11.0 -0500
+++ term.c 2007-02-15 16:28:24.0 -0500
@@ -78,9 +78,12 @@
char buffer[512] = ;
int termfd = -1, parseokay = 0, i;
+ if (!option_console || !option_nofork)
+ term = xterm;
+
if (!term)
return 0;
- if (!option_console || option_nocolor || !option_nofork)
+ if (option_nocolor)
return 0;
for (i=0 ;; i++) {
/*- end -*/
--Michel
___
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Re: [asterisk-users] Call forwarding
With the call forward button on the phone? ;) PaulH Stefan it didnt help :( Is there is other way to implement call forwarding? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Long call setup times on SIP to zaptel calls
All of our SIP phones dial instantly when the users finished dialing. We can do this because we have no ambiguous extension lengths. i.e. no _XXX and _ and we don't use the . pattern match. Shane Spencer wrote: I only say this because nobody in our office knew how to use the checkmark on snom phones to initiate a call, they always just waited for the phone to initiate the call for them :) On 2/15/07, Shane Spencer [EMAIL PROTECTED] wrote: do your sip phones dial after a timeout? If the timeout is set to around 5 seconds you may have a dialplan issue on your sip hardware. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect PBX CO Port to TDM FXS Port
Daniel Kocher wrote: Daniel Kocher wrote: I would like to connect a Legacy PBX (Avaya IP Office 406) to an Asterisk Server. The Avaya has 3 CO Ports available. I thought buying a TDM30B card with 3 FXS ports to connect the * to the Avaya CO Ports. Is this the right approach? Does any one have experience with such a configuration? Thanks in advance for all recommandations and suggestions Daniel Kocher It kind of depends on what you're trying to accomplish. What do you want to be able to do with this connection? -Dave I would like to use the * as VoIP Gateway. Something like that: A user takes off a phone on a Avaya extension and dials for example 8 to reach the CO Port. Then Asterisk answers and sends a dial tone. The user dails a numer and Asterisk is doing the rest! (Sending the call to an SIP or IAX Provider) Yes, it works fine like that. We have several systems using * as a gateway. Jorge Mendoza ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New AstLinux Branch: RT PREEMPT (realtime Linux) - Looking for testers
Hello everyone, Now that astlinux-trunk has been coming along very nicely, I thought I would try to add support for hard realtime capabilities to AstLinux. If everything works (and there are no problems with zaptel), with a little tweaking this should improve the audio quality on systems with high loads (and probably any system at that) - especially if it is finely tuned and has zaptel cards. The stats: - Asterisk 1.2.14 - Zaptel 1.2.13 - Kernel 2.6.20 - rt patch 2.6.20-rt5 - everything else from AstLinux... If you would like to hack on this, give the astlinux rt branch a try: svn co https://astlinux.svn.sourceforge.net/svnroot/astlinux/branches/rt astlinux-rt If you just want to try it on something, I made a bootable iso (make iso from the devel environment). Get it here: http://www.krisk.org/astlinux/astlinux-rt-r588.iso (sorry about the krisk.org domain - I don't feel like dealing with SourceForge right now) Further reading: http://rt.wiki.kernel.org/index.php/RT_PREEMPT_HOWTO http://people.redhat.com/mingo/realtime-preempt/ http://rlove.org/schedutils/ I'm looking for any and all suggestions from Asterisk code gurus - what things can we do in Asterisk/Zaptel to maximize the potential when running with RT PREEMPT? Thanks, I look forward to hearing what everyone has to say. -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] The High Performance Echo Canceller (HPEC)
How do you fake echo for testing purposes then? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nic Bellamy Sent: Thursday, 15 February 2007 3:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC) Wireless wrote: Does anyone know if the HPEC will work on a Sangoma A200 / 2 port FXO card? (I'm assuming so as it still uses Zapel) I've 2 PSTN lines one of which I cannot get rid of the echo, I've tried a 2GHz machine as apposed to my normal P3 650MHz and this made no difference. Would the 650Mhz be enough to run HPEC on one line (I assume only needing one licence) It should work, providing all the Wanpipe stuff is ready to work with Zaptel 1.2.13. As far as performance, you should be able to get one, maybe two channels of 1024 tap cancellation on the P3, but I'd advise careful testing, perhaps even using oprofile for a while to keep an eye on what's using what. You also have to watch out extra carefully due to the following: HPEC works in sparse mode, meaning it can cover 1024 taps, but just cancels echo in the parts where there is echo - hence CPU usage will likely change quite a bit with different echo paths - ie. a simple single reflection path will use less CPU than a complicated path with more than one reflection. Cheers, Nic. -- Nic Bellamy, Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
