[asterisk-users] How to enter bridge_native_loop???
Hi, I am using asterisk-1.4.0. I am inquisitive of what Packet2Packet bridge (bridge_p2p_loop) does and what Native bridge (bridge_native_loop) does. I have configured my dial plans and options such that I can enter bridge_p2p_loop. However, I am unable to enter bridge_native_loop for some reason. I have the following extensions: exten = 7126,1,Dial(SIP/lin_santosh) exten = 7126,s+1,Hangup exten = 7140,1,Dial(SIP/win_test) exten = 7140,s+1,Hangup My sip.conf is as: [lin_santosh] type=friend regexten=7126 callerid=LIN Santosh 7126 host=dynamic nat=yes canreinvite=no allow=all [win_test] type=friend regexten=7140 callerid=WIN Test 7140 host=dynamic nat=yes canreinvite=no allow=all My features.conf has: [featuremap] ;blindxfer = # ; Blind transfer (default is #) ;disconnect = *0 ; Disconnect (default is *) ;automon = *1 ; One Touch Record a.k.a. Touch Monitor ;atxfer = *; Attended transfer ;parkcall = #72; Park call (one step parking) Should this not be enough for bridging to happen via bridge_native_loop? Am I missing something? Thanks in anticipation. Regards, Santosh. ___ --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] Hinting and Realtime
8 mar 2007 kl. 14.36 skrev René Enskat: hello all, My problem if i have my extensions and sipusers in a realtime database it is not possible to use BLF or hinting. i see only idle or unavailable status but if the phone is ringing or in use i can't see it. Is there a fix or any workaround? Version is Release 1.4.1 The whole idea with realtime objects in SIP is not to keep them in memory. We release them when we don't need them and reload them when needed. That's why realtime objects lack a lot of functionality compared with static objects. /O___ --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] Packet2Packet Bridging Questions
8 mar 2007 kl. 21.05 skrev Daryl Jurbala: OK...that makes much more sense. So here's my follow-up question: what's the easiest way to check if I'm native bridging a call. I'm trying to offload as much RTP traffic as possible, and want to have a way to check quickly (there are well over 50 calls on each of these boxes at any given time). I've been going the ethereal route, which is great for debugging, but not so good for a quick look. During the call, run sip show channel xxxyyzz and check Audio IP. If the audio IP doesn't belong to your Asterisk server, media is handled through the native bridge. /O ___ --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] How to enter bridge_native_loop???
9 mar 2007 kl. 08.52 skrev Santosh Raghuram: Hi, I am using asterisk-1.4.0. I am inquisitive of what Packet2Packet bridge (bridge_p2p_loop) does and what Native bridge (bridge_native_loop) does. I have configured my dial plans and options such that I can enter bridge_p2p_loop. However, I am unable to enter bridge_native_loop for some reason. I have the following extensions: exten = 7126,1,Dial(SIP/lin_santosh) exten = 7126,s+1,Hangup exten = 7140,1,Dial(SIP/win_test) exten = 7140,s+1,Hangup My sip.conf is as: [lin_santosh] type=friend regexten=7126 callerid=LIN Santosh 7126 host=dynamic nat=yes canreinvite=no allow=all You have set canreinvite to no, thus disabling native briding. /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * Asterisk SIP Masterclass - Stockholm, Sweden, May 7-11 ___ --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] RE: Zaptel problem after upgrading to 1.2.16
Further to this, I believe that my problem is that I'm also now running udev. When I compiled and installed Zaptel, I did the make install-udev step, however the permissions in my udev directory don't look correct. I am running Asterisk as root (this is on a debian system btw), but this is the content of my zaptel.rules file. # zaptel devices with ownership/permissions for running as non-root KERNEL==zapctl, NAME=zap/ctl, OWNER=asterisk, GROUP=asterisk, MODE=0660 KERNEL==zaptranscode, NAME=zap/transcode, OWNER=asterisk, GROUP=asterisk, MODE=0660 KERNEL==zaptimer, NAME=zap/timer, OWNER=asterisk, GROUP=asterisk, MODE=0660 KERNEL==zapchannel, NAME=zap/channel, OWNER=asterisk, GROUP=asterisk, MODE=0660 KERNEL==zappseudo, NAME=zap/pseudo, OWNER=asterisk, GROUP=asterisk, MODE=0660 KERNEL==zap[0-9]*, NAME=zap/%n, OWNER=asterisk, GROUP=asterisk, MODE=0660 zaptel.rules (END) What do I need to do for Asterisk to be able to see the Zaptel device while running as root? Again, any help is much appreciated. Regards, Mark. From: Mark Davies Sent: Friday, 9 March 2007 4:13 PM To: 'asterisk-users@lists.digium.com' Subject: Zaptel problem after upgrading to 1.2.16 Hi guys, I'm hoping I've made a silly mistake here, but I've been staring at the screen for the past few hours and I can't work it out. I upgraded to 1.2.16 recently, and am having problems with zaptel. The card is detected, I get a reasonable output from ztcfg -vv, and zttool shows the installed module (TDM400) with one FXS module. But when I start asterisk, I get an error saying that my IAX connection won't work in trunked mode because there's no timing interface. Zaptel doesn't show up in the output of show channeltypes. Should there be a problem with using the trunk version of zaptel, but 1.2.16 of asterisk? Are there any places that I can specifically load/enable the zaptel module? Any help much appreciated before I go insane... J Regards, Mark. ___ --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 problem after upgrading to 1.2.16
On Fri, Mar 09, 2007 at 04:13:04PM +0900, Mark Davies wrote: Hi guys, I'm hoping I've made a silly mistake here, but I've been staring at the screen for the past few hours and I can't work it out. I upgraded to 1.2.16 recently, and am having problems with zaptel. The card is detected, I get a reasonable output from ztcfg -vv, and zttool shows the installed module (TDM400) with one FXS module. Can you use those modules from Asterisk? Do you get a dialtone on the phone? But when I start asterisk, I get an error saying that my IAX connection won't work in trunked mode because there's no timing interface. Zaptel doesn't show up in the output of show channeltypes. Is there any? Is chan_zap.so loaded ? Do you even have /usr/lib/asterisk/modules/chan_zap.so ? If yes: do you have a valid timing source: try zttest . Should there be a problem with using the trunk version of zaptel, but 1.2.16 of asterisk? Did the same configuration work with other versions of Asterisk 1.2? What version of Zaptel do you have? Are there any places that I can specifically load/enable the zaptel module? Do you have automatic modules loading enabled in modules.conf? -- 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] RE: Zaptel problem after upgrading to 1.2.16
On Fri, Mar 09, 2007 at 06:06:01PM +0900, Mark Davies wrote: Further to this, I believe that my problem is that I'm also now running udev. When I compiled and installed Zaptel, I did the make install-udev step, however the permissions in my udev directory don't look correct. I am running Asterisk as root (this is on a debian system btw), but this is the content of my zaptel.rules file. You don't need install-udev there: grep zap /etc/udev/rules.d/* Just remove your file. Add asterisk to the group 'dialout' or edit 020_permissions.rules to use have: SUBSYSTEM==zaptel, USER=asterisk instead of: SUBSYSTEM==zaptel, GROUP=dialout The default mode is already 0660, so no need to set it explicitly. I recommend that you remove /etc/udev/rules.d/zaptel , as I'm not sure if it actually gets activated. # zaptel devices with ownership/permissions for running as non-root KERNEL==zapctl, NAME=zap/ctl, OWNER=asterisk, GROUP=asterisk, MODE=0660 KERNEL==zaptranscode, NAME=zap/transcode, OWNER=asterisk, GROUP=asterisk, MODE=0660 KERNEL==zaptimer, NAME=zap/timer, OWNER=asterisk, GROUP=asterisk, MODE=0660 KERNEL==zapchannel, NAME=zap/channel, OWNER=asterisk, GROUP=asterisk, MODE=0660 KERNEL==zappseudo, NAME=zap/pseudo, OWNER=asterisk, GROUP=asterisk, MODE=0660 KERNEL==zap[0-9]*, NAME=zap/%n, OWNER=asterisk, GROUP=asterisk, MODE=0660 zaptel.rules (END) What do I need to do for Asterisk to be able to see the Zaptel device while running as root? Asterisk running as root? Bad! But in the case the permissions on the udev files don't really matter. -- 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] Which VoIP router and switch to use for medium size business
On Fri, 9 Mar 2007, Zeeshan Zakaria wrote: Hi everybody, What is a proper setup for a medium size business with about 20 IP phones and 20 computers. Right now they are using a regular Linksys router which we use at homes. Their switch is also a very standard switch. Now they need to put there something better and VoIP compatible. What people use out there in serious and professional VoIP installations for medium size businesses? Is there a good 24 port router with VoIP compatibility with no need of an extra switch? Please advice me for all the equipment I'd need for a complete network upgrade. You'd need to supply more details for a detailled answer, (like what's the budget ;-) and why do you think you need something better? What do you currently have, and are they actually having problems at present, or it is just a percieved problem? But if this was me, and it was a green site installation, and money was a bit of a consideration (it usually is IME for small businesses unless they are new, VC funded with millions in the bank and Porsches in the car park ;-), So I'd start with 2 decent enough 24-port Ethernet switches and run at least 2 Ethernet lines to each desk (one for the PC, one for the phone) back to a patch panel (actually, these days, I run 4 to each desk, but a lot depends on the type of company!) You might want to investigate switches with PoE capabilities - in an ideal world this would be good, but it adds to the expense, and watch out for their power carrying ability - some can only power 8 of 16 sockets for example - 20 phones at 6W each is 120W which is quite a bit to add to the PSU capacity of a little 1U Ethernet switch! Some switches can let use use 15W per port, for 8 ports, or 7.5W for 16 ports - the Grandstream phones I use claim to suck no more than 6W - is that lucky or what ;-) I'd steer away from using the in-line connections that many IP phones have - unless you were desperately short of cabling capabiltiy, or money. (but I have used the switch facilities on Grandstream GXP phones in an small office environment where I didn't have much choice and not had any issues with it. (Although if you reboot or unplug the phone, it takes the PC offline!) Make sure the switch in the phone is a switch if you need to use it! In the Grandstream Budgetone 100's it's a 10Mb HUB, not a 10/100 switch!) Then plug all phones and the asterisk box into one switch and all PCs and servers into the other. You can then plug each switch into the router, if it has a switch of it's own, (a lot of the netgear ADSL routers have built-in 4-port switches) or have a small 3rd top-level switch to connect the 2 switches and rotuer together. or you can daisy-chain the switches into the router if it only has one port. So without doing anything special, this will keep VoIP traffic inside one switch and PC/PC/Server traffic inside the other and by the switching nature of the switches, stop traffic from PC to PC/Server interfering with switched VoIP traffic on the other switch. If you can't afford the luxury of separate Ethernet switches, then you might need to look into something a bit more exotic and use Layer 2 QoS/801.p/VLan services, etc. The above isn't perfect, but for your average small office, it's hard to beat for a price. Things can go wrong though - someone plugging a PC into the VoIP switch, then running network traffic intensive apps to other PCs or servers (games, viruses). Broadcast/APR storms but these are rare these days (however if you want to experiment, loopback 2 ethernet ports and stand well back!) Some switches will detect this and suppress excessive ARP broadcast traffic though. Your router choice will depends on what you are trying to achieve - if you are placing calls over the Internet, then you might want a router which has QoS functions - however, the reality is that you can only effectively use QoS when you control every aspect of the link - which with most ISPs you can't. (And you don't say if you have an ADSL, Cable, Leased line, etc. connection - not that it matters that much, however) Most routers will make a good effort though, but over the big bad Internet (and incoming data to your site in particular) you have no control over. Saying that, with a good ISP and reasonable staff, most of the time you get away with it, and I regularly chat with my clients and friends over the 'net even when I know some of them have no QoS at all on their company routers. One site in particular has 100 staff, a 4Mb Internet line and I regularly make calls over their non QoS'd Internet line to other sites without any issues at all. (But it just takes one person running some agressive P2P software to kill it for everyone!) If you are running VPNs to other sites, make sure the router is up to it! After many years of using Drayteks, I now find them a PITA as they can only sustain about 1.5Mb/sec through an encrypted VPN and with
Re: [asterisk-users] Boot order of 2 TE110P and 1 TDM400P in the same machine
On Thu, Mar 08, 2007 at 05:01:15PM -0800, Jose Bertuzzi wrote: Hello Everyone, I checked with zttool that sometimes after the machine boots the order of the boards is changed like this: â Alarms Span â OK Digium Wildcard TE110P T1/E1 Card 0 â OK Digium Wildcard TE110P T1/E1 Card 1 â OK Wildcard TDM400P REV I Board 1 and sometimes: â Alarms Span â OK Wildcard TDM400P REV I Board 1â OK Digium Wildcard TE110P T1/E1 Card 1 â OK Digium Wildcard TE110P T1/E1 Card 0 What do I have to configure in order to the boards appear in the same position and the configuration work always?? Which linux distribution? -- 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] Is there any variable for Voicemail Password in Asterisk
Hi guys This is my Ist post on this group. Is there any variable like ($VM_CALLERID for voicemail mailbox) for accessing Asterisk Voicemail password which is set through comedian mail.?? plz reply me as soon as possible htmldivPRE class=quoteIMG height=2 src=http://graphics.hotmail.com/greypixel.gif; width=100% vspace=9/PREPRE class=quoteFONT face=Courier New, Courier, Monospace color=#33 size=2Syed Jamshed Zaidi BRAsterisk Admin/Developer BR@ Axvoice +92-0321-4087492 BR(JAMY-VIRUS)/FONT/PREPRE class=quotePRE class=quoteFONT face=Courier New, Courier, Monospace color=#33 size=2Shoot for the moon. Even if you miss, you'll land among the stars/FONT/PRE/PRE/div/html _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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] Re: Back to back E1 - asterisk = toshiba pbx - Call droping issue [ ref:00D36mPe.50032wycQ:ref ]
Hi All, Thanks for every one who helped me on this regard. I think i was able to rictify the problem. what i did is remove callprogress=yes usecallinpres=yes and restart asterisk. Today i didn't report any drop calls. Many thanks for Eric. :) I hope this situation will continue. Regards, Vidura. On 3/8/07, Vidura Senadeera [EMAIL PROTECTED] wrote: Hi, Opps ...there are some more attachments i missed to send you. Please refer. sorry for the inconvenience occured. Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk On 3/8/07, Vidura Senadeera [EMAIL PROTECTED] wrote: Hi, Today I have reported 10 calls drop within 2.5 hours period of time. This is being a huge issue. I'm using Asterisk 1.2.14 and zaptel 1.2.12 and libpri-1.2.4. Pls find attached files you have requested. Thanks, Vidura. On 3/8/07, Digium Support [EMAIL PROTECTED] wrote: Hi, Please make sure you are running the latest 1.2 or 1.4 stable releases of Asterisk/Zaptel and Libpri. Also could you send me the output of these two commands: cat /proc/interrupts lspci -bv Please let us know if you have any questions. -Regards David Faulk Digium Support Technician Digium Certified Asterisk Professional Digium, Inc. 150 West Park Loop, Suite 100 Huntsville, AL 35806 +1-256-428-6000 www.digium.com ref:00D36mPe.50032wycQ:ref -- Thanks Regards, Vidura B. Senadeera. -- Thanks Regards, Vidura B. Senadeera. -- Thanks Regards, Vidura B. Senadeera. ___ --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] Queue announcing hold sequence instead of hold time
From: Drew Gibson Hi, We recently updated from an early Asterisk 1.2 SVN to 1.2.15 (on Debian Sarge) and the behaviour of our Call Centre queues has changed slightly. Before the upgrade, when a caller was waiting in the queue, the estimated hold time was announced as expected (estimated hold time is less than 2 minutes ...). Now the caller gets an announcement of their sequence in the queue (Your call is now first in line ...). I believe that the only changes I have made to queues.conf and agents.conf is the addition of the context= statement and editing the list of agents. Has anyone else seen this? What am I missing? regards, Drew Drew, This has been normal behaviour for as long as I can remember. The caller hears the estimated time until they are next in line, then they hear the 'next in line' announcement. Sincerely, Trevor Hammonds ___ --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] Queue announcing hold sequence instead of hold time
Trevor, I also have the same problem as Drew, and that isn't how mine works. Even though I told it to announce the time, I get the first in line as well as second in line. I've tested it up to 5 people sitting in the queue line, and each gets the same message (space, not time). Rob Trevor G. Hammonds wrote: From: Drew Gibson Hi, We recently updated from an early Asterisk 1.2 SVN to 1.2.15 (on Debian Sarge) and the behaviour of our Call Centre queues has changed slightly. Before the upgrade, when a caller was waiting in the queue, the estimated hold time was announced as expected (estimated hold time is less than 2 minutes ...). Now the caller gets an announcement of their sequence in the queue (Your call is now first in line ...). I believe that the only changes I have made to queues.conf and agents.conf is the addition of the context= statement and editing the list of agents. Has anyone else seen this? What am I missing? regards, Drew Drew, This has been normal behaviour for as long as I can remember. The caller hears the estimated time until they are next in line, then they hear the 'next in line' announcement. Sincerely, Trevor Hammonds ___ --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] RE: Coaching in asterisk
Fix on an agent? What do you mean? We make call center software and our clients usually have 200 to more than a thousand agents, and some agents are even working from a remote location like their homes. We have a small application for the supervisor throught which he/she can view the status of all the agents currently on the system and by clicking a button on the agent name, he/she can monitor/coach any one of them he/she wish. We are add Asterisk support now, and I want this feature to be supported. -Original Message- From: [EMAIL PROTECTED] on behalf of Steve Totaro Sent: Thu 3/8/2007 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Coaching in asterisk Must be a quiet and small call center without high cubicle walls. There is no way that would be an issue at the call center I setup. 16 agents to a team and all of them on the phone all the time, you cannot even fix in on an agent if you wanted to, there was too much noise. Thanks, Steve Doug Garstang wrote: We used ChanSpy to allow a supervisor to listen in on the calls of their staff. There was one huge problem with this, which I imagine would affect whisper as well. The supervisor typically sat fairly close to the worker, and could hear both the voice of the worker as they spoke AND the delayed voice coming through their head phones. It was rather distracting and made it difficult to really be practical. Doug. Dean Collins wrote: Yep, it's called Whisper Check in voip-info.org I think I've read stuff about it there. 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 Wai Wu Sent: Thursday, 8 March 2007 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Coaching in asterisk Is there a way to setup a conference where party A can coach another Party B, at the same time, all other parties cannot hear party A? In order words, partis A and B can hear every one, and party A can only be heard by party B. Thnx ___ --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 attachment: winmail.dat___ --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] disable client side hangup after dialing 911
Hi Has anyone tried to reproduce the following behavior that a standard phone line does with 911. Normally if someone calls 911 and hangs up after the call has been established then the line is not dropped because it is held by the 911 agent. If you pickup your phone you should still be connected to the 911 agent and be able to talk to him. The call is dropped only when the 911 agent hangs up on his side. Is there a way in asterisk to disable the hangup from the client after he has dialed 911 ? Or maybe asterisk can keep the channel up and call back the user to re-establish the call until the hangup comes from the other side Thanks Patrick ___ --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 load balancing
What kind of hardware are you using in your setup? I'm using dell GX150 PIII 1G 512M, because I can get them in quantity and the parts are easily interchangeable Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Thursday, March 08, 2007 6:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call load balancing On Thu, 8 Mar 2007, David Ruggles wrote: I've got a system I'm putting together to handle IVR calls with * I have one head system that terminates two PRIs. It routes the calls from the PRIs to * boxes using IAX I'm planning on having four or five * boxes. The * boxes run AGI scripts to process the IVR calls. Can I load balance the routing if I have five calls each of the IVR * boxes gets two call and the next call would go to the system that currently has the lowest number of calls? Quick answer, yes. How is more interesting :) First, unless your AGI's are massive or incredibly inefficient, 2 PRI's won't swamp your IVR boxes. I have 3 1u servers each with 2 PRI's forwarding all 138 calls to a single application server. All of the PRI's could be handled by 1 1u but management wanted flexibility and redundancy. The application server does IVR, conferencing, records messages, plays canned stories, credit card processing, etc, etc, etc. All implemented with a bunch of AGI's written in C. Each call executes a minimum of 9 AGI's and yes, some AGI consolidation is planned. All database work is handled by a separate box. Anyway, back to your question, how about your head system running an AGI that connects to the manager interface on the IVR boxes to find out how many calls each is currently processing? You could set a channel variable with the least busy host name and use that in your dial statement. If you passed the IVR host name list to the AGI, you could take a box out of service by editing and reloading your dialplan. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] disable client side hangup after dialing 911
Two things. 1) This is a bug(feature) of standard analog switchs which only clear the talk path when both sides of the call are terminated. 2) You should post this in the asterisk development list. -Original Message- From: [EMAIL PROTECTED] on behalf of Patrick Fortin Sent: Fri 3/9/2007 9:22 AM To: asterisk-users-lists.digium.com Subject: [asterisk-users] disable client side hangup after dialing 911 Hi Has anyone tried to reproduce the following behavior that a standard phone line does with 911. Normally if someone calls 911 and hangs up after the call has been established then the line is not dropped because it is held by the 911 agent. If you pickup your phone you should still be connected to the 911 agent and be able to talk to him. The call is dropped only when the 911 agent hangs up on his side. Is there a way in asterisk to disable the hangup from the client after he has dialed 911 ? Or maybe asterisk can keep the channel up and call back the user to re-establish the call until the hangup comes from the other side Thanks Patrick ___ --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 attachment: winmail.dat___ --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] Is Allison going to be banned from foreign travel over polar bears?
Steve Prior wrote: I read this story and thought of Allison's prompt to try not to think about blue eyed polar bears. Will she be banned from foreign travel now? Steve Prior -- snip -- WASHINGTON (Reuters) - Polar bears, sea ice and global warming are taboo subjects, at least in public, for some U.S. scientists attending meetings abroad, environmental groups and a top federal wildlife official said on Thursday. Foreign travel? Do you mean, like trips to the US? regards, Drew ___ --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] disable client side hangup after dialing 911
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 If you're using all Zaptel channels for the call, it sounds like you want operator services mode (Dial command flag). O([x]) - Operator Services mode (Zaptel channel to Zaptel channel only, if specified on non-Zaptel interface, it will be ignored). When the destination answers (presumably an operator services station), the originator no longer has control of their line. They may hang up, but the switch will not release their line until the destination party hangs up (the operator). Specified without an arg, or with 1 as an arg, the originator hanging up will cause the phone to ring back immediately. With a 2 specified, when the operator flashes the trunk, it will ring their phone back. Patrick Fortin wrote: Hi Has anyone tried to reproduce the following behavior that a standard phone line does with 911. Normally if someone calls 911 and hangs up after the call has been established then the line is not dropped because it is held by the 911 agent. If you pickup your phone you should still be connected to the 911 agent and be able to talk to him. The call is dropped only when the 911 agent hangs up on his side. Is there a way in asterisk to disable the hangup from the client after he has dialed 911 ? Or maybe asterisk can keep the channel up and call back the user to re-establish the call until the hangup comes from the other side Thanks Patrick ___ --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 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF8XSaRhLSniguQyERAlLUAKDYAWXZZA1Ol8MwhjeCfJC4J95DaQCcDls9 VZNAPSItxLmyHo37Xji1CaE= =fP9l -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] Newbie Question
[test] disallow=all allow=gsm ;GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw Are you sure that the xlite phone can handle gsm?? ___ --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 load balancing
telco servers are Supermicro 2.8GHz P4, 1GB, Digium te410p with 2 PRI plugged in. application server is HP DL380 3.06GHz Xeon x2 (4 cores), 3GB, Digium te410p (timing only, all calls over IAX) database server is HP DL380 3.06GHz Xeon x2 (4 cores), 3GB No failures in over 2 years. On Fri, 9 Mar 2007, David Ruggles wrote: What kind of hardware are you using in your setup? I'm using dell GX150 PIII 1G 512M, because I can get them in quantity and the parts are easily interchangeable Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Thursday, March 08, 2007 6:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call load balancing On Thu, 8 Mar 2007, David Ruggles wrote: I've got a system I'm putting together to handle IVR calls with * I have one head system that terminates two PRIs. It routes the calls from the PRIs to * boxes using IAX I'm planning on having four or five * boxes. The * boxes run AGI scripts to process the IVR calls. Can I load balance the routing if I have five calls each of the IVR * boxes gets two call and the next call would go to the system that currently has the lowest number of calls? Quick answer, yes. How is more interesting :) First, unless your AGI's are massive or incredibly inefficient, 2 PRI's won't swamp your IVR boxes. I have 3 1u servers each with 2 PRI's forwarding all 138 calls to a single application server. All of the PRI's could be handled by 1 1u but management wanted flexibility and redundancy. The application server does IVR, conferencing, records messages, plays canned stories, credit card processing, etc, etc, etc. All implemented with a bunch of AGI's written in C. Each call executes a minimum of 9 AGI's and yes, some AGI consolidation is planned. All database work is handled by a separate box. Anyway, back to your question, how about your head system running an AGI that connects to the manager interface on the IVR boxes to find out how many calls each is currently processing? You could set a channel variable with the least busy host name and use that in your dial statement. If you passed the IVR host name list to the AGI, you could take a box out of service by editing and reloading your dialplan. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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 Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] disable client side hangup after dialing 911
On 3/9/07, Wai Wu [EMAIL PROTECTED] wrote: Two things. 1) This is a bug(feature) of standard analog switchs which only clear the talk path when both sides of the call are terminated. Well, not exactly. The call will not terminated until the caller (not both) hangs up. I don't knew the american emergency numbers, but in europe, if the caller hangs up, the call will terminated. If the called party hangs up, the call will not terminated. ___ --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 problem after upgrading to 1.2.16
Mark Davies wrote: Hi guys, I'm hoping I've made a silly mistake here, but I've been staring at the screen for the past few hours and I can't work it out. I upgraded to 1.2.16 recently, and am having problems with zaptel. The card is detected, I get a reasonable output from ztcfg -vv, and zttool shows the installed module (TDM400) with one FXS module. But when I start asterisk, I get an error saying that my IAX connection won't work in trunked mode because there's no timing interface. Zaptel doesn't show up in the output of show channeltypes. Should there be a problem with using the trunk version of zaptel, but 1.2.16 of asterisk? Are there any places that I can specifically load/enable the zaptel module? Is there a chan_zap.so in /usr/lib/asterisk/modules? Are you upgrading over an existing install of Asterisk, or did you remove the old ver first? I had trouble building Asterisk 1.2.15 with the 1.4 zaptel driver. Asterisk did not find the zaptel src and wouldn't build chan_zap.so. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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] Call load balancing
That's cool, but I doubt my systems could handle that same load ;) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Friday, March 09, 2007 9:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Call load balancing telco servers are Supermicro 2.8GHz P4, 1GB, Digium te410p with 2 PRI plugged in. application server is HP DL380 3.06GHz Xeon x2 (4 cores), 3GB, Digium te410p (timing only, all calls over IAX) database server is HP DL380 3.06GHz Xeon x2 (4 cores), 3GB No failures in over 2 years. On Fri, 9 Mar 2007, David Ruggles wrote: What kind of hardware are you using in your setup? I'm using dell GX150 PIII 1G 512M, because I can get them in quantity and the parts are easily interchangeable Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Thursday, March 08, 2007 6:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call load balancing On Thu, 8 Mar 2007, David Ruggles wrote: I've got a system I'm putting together to handle IVR calls with * I have one head system that terminates two PRIs. It routes the calls from the PRIs to * boxes using IAX I'm planning on having four or five * boxes. The * boxes run AGI scripts to process the IVR calls. Can I load balance the routing if I have five calls each of the IVR * boxes gets two call and the next call would go to the system that currently has the lowest number of calls? Quick answer, yes. How is more interesting :) First, unless your AGI's are massive or incredibly inefficient, 2 PRI's won't swamp your IVR boxes. I have 3 1u servers each with 2 PRI's forwarding all 138 calls to a single application server. All of the PRI's could be handled by 1 1u but management wanted flexibility and redundancy. The application server does IVR, conferencing, records messages, plays canned stories, credit card processing, etc, etc, etc. All implemented with a bunch of AGI's written in C. Each call executes a minimum of 9 AGI's and yes, some AGI consolidation is planned. All database work is handled by a separate box. Anyway, back to your question, how about your head system running an AGI that connects to the manager interface on the IVR boxes to find out how many calls each is currently processing? You could set a channel variable with the least busy host name and use that in your dial statement. If you passed the IVR host name list to the AGI, you could take a box out of service by editing and reloading your dialplan. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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 Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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[2]: [asterisk-users] Boot order of 2 TE110P and 1 TDM400P in the same machine
I got the same thing on a Ubuntu Dapper. -Original Message- From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Cc: Sent: Fri, 9 Mar 2007 12:12:45 +0200 Delivered: Fri, 09 Mar 2007 06:45:09 Subject:[asterisk-users] Boot order of 2 TE110P and 1 TDM400P in the same machine On Thu, Mar 08, 2007 at 05:01:15PM -0800, Jose Bertuzzi wrote: Hello Everyone, I checked with zttool that sometimes after the machine boots the order of the boards is changed like this: â Alarms Span â OK Digium Wildcard TE110P T1/E1 Card 0 â OK Digium Wildcard TE110P T1/E1 Card 1 â OK Wildcard TDM400P REV I Board 1 and sometimes: â Alarms Span â OK Wildcard TDM400P REV I Board 1â OK Digium Wildcard TE110P T1/E1 Card 1 â OK Digium Wildcard TE110P T1/E1 Card 0 What do I have to configure in order to the boards appear in the same position and the configuration work always?? Which linux distribution? -- 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 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=1,1173435807.682905.10862.almora.hst.terra.com.br,4630,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
Re: [asterisk-users] Queue announcing hold sequence instead of hold time
Trevor G. Hammonds wrote: From: Drew Gibson Hi, We recently updated from an early Asterisk 1.2 SVN to 1.2.15 (on Debian Sarge) and the behaviour of our Call Centre queues has changed slightly. Before the upgrade, when a caller was waiting in the queue, the estimated hold time was announced as expected (estimated hold time is less than 2 minutes ...). Now the caller gets an announcement of their sequence in the queue (Your call is now first in line ...). I believe that the only changes I have made to queues.conf and agents.conf is the addition of the context= statement and editing the list of agents. Has anyone else seen this? What am I missing? regards, Drew Drew, This has been normal behaviour for as long as I can remember. The caller hears the estimated time until they are next in line, then they hear the 'next in line' announcement. Sincerely, Trevor Hammonds Hi Trevor, I should have given a better example. Like Rob Schall, the 2nd, 3rd, 4th, etc callers all get a sequence number rather than an estimated time. Rob, are there any common elements in our configs, like t or H options that might be getting in the way? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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
[asterisk-users] sip tunnel
Dears my Internet Provider , prevents , sip connections, between sip client(sip phone) and sip server, (asterisk + ser) . both of client and server are mine. is there any solution for tunneling the sip packets? best Mani Don't pick lemons. See all the new 2007 cars at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ --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] Queue announcing hold sequence instead of hold time
All - Next step here would probably be to open a bug on bugs.digium.com with a full VERBOSE/DEBUG log along with associated config files so we can troubleshoot this and fix it if there's a problem. Thanks. On 3/9/07, Drew Gibson [EMAIL PROTECTED] wrote: Trevor G. Hammonds wrote: From: Drew Gibson Hi, We recently updated from an early Asterisk 1.2 SVN to 1.2.15 (on Debian Sarge) and the behaviour of our Call Centre queues has changed slightly. Before the upgrade, when a caller was waiting in the queue, the estimated hold time was announced as expected (estimated hold time is less than 2 minutes ...). Now the caller gets an announcement of their sequence in the queue (Your call is now first in line ...). I believe that the only changes I have made to queues.conf and agents.conf is the addition of the context= statement and editing the list of agents. Has anyone else seen this? What am I missing? regards, Drew Drew, This has been normal behaviour for as long as I can remember. The caller hears the estimated time until they are next in line, then they hear the 'next in line' announcement. Sincerely, Trevor Hammonds Hi Trevor, I should have given a better example. Like Rob Schall, the 2nd, 3rd, 4th, etc callers all get a sequence number rather than an estimated time. Rob, are there any common elements in our configs, like t or H options that might be getting in the way? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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 -- Bird's The Word Technologies, Inc. http://www.btwtech.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
[asterisk-users] YAACID and manager.conf security
Hi - I am going to open port 5038 on my firewall so that I can use YAACID to spawn browser popups on an incoming call. My question is, under manager.conf, what are the suggested settings so that I can get the browser popups only? I'll be at different IPs so I can't lock it down that way.. I guess I don't need any write access? [managername] secret=secretword read=system,call,log,verbose,command,agent,user write=system,call,log,verbose,command,agent,user thanks Todd ___ --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] Boot order of 2 TE110P and 1 TDM400P in the same machine
On Fri, Mar 09, 2007 at 12:18:00PM -0300, Melcon Moraes wrote: I got the same thing on a Ubuntu Dapper. On Ubuntu and Debian, put your modules in the desired order in /etc/modules . And just in case you need to unload the module and load them again, the asterisk init.d script in the Debian package does this when you call its asterisk-fix target: /etc/init.d/asterisk stop /etc/init.d/zaptel unload # unload all zaptel modules /etc/init.d/module-init-tools start # load modules from /etc/modules /etc/init.d/zaptel start# run ztcfg etc. /etc/init.d/asterisk start -- 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] sip tunnel
try changing bindport of asterisk from 5060 to something else . On 09/03/07, Pezhman Lali [EMAIL PROTECTED] wrote: Dears my Internet Provider , prevents , sip connections, between sip client(sip phone) and sip server, (asterisk + ser) . both of client and server are mine. is there any solution for tunneling the sip packets? best Mani Don't pick lemons. See all the new 2007 cars at Yahoo! Autos. http://autos.yahoo.com/new_cars.html ___ --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 load balancing
Anyway, back to your question, how about your head system running an AGI that connects to the manager interface on the IVR boxes to find out how many calls each is currently processing? You could set a channel variable with the least busy host name and use that in your dial statement. If you passed the IVR host name list to the AGI, you could take a box out of service by editing and reloading your dialplan. Can you give me a link to more information about how to use the management interface? I've been having a hard time trying to track down more advanced documentation and reference material. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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
Re: [asterisk-users] Issues with a Linksys SPA 2102 and asterisk
Any gurus out there with experience with the SPA 2102 against asterisk 1.2.14? They work fine with Asterisk; most likely it's your wireless link that's the cause of your problem. The jitter buffer will only affect received audio, i.e. on your side, and since that is fine, you probably don't need to adjust it. Instead try this: 1) Change packet size in increments of 20 ms (i.e. 0.02, 0.04 or perhaps 0.06). Your wireless link may not like too many small packets. 2) Turn off silence suppression if it's on. 3) Try a different codec -- g726-32 or even ulaw to see if it makes a difference. See if that helps. --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
[asterisk-users] Another Faxing Question
This probably came up before, but I have a faxing question for everyone.
I have a simple extension setup to use rxfax to receive faxes sent to
asterisk. It is:
exten = s,1,Answer()
exten = s,n,AbsoluteTimeout(300)
exten =
s,n,Set(FAXFILE=/var/spool/asterisk/fax/${ARG1}_${CALLERIDNUM}_${UNIQUEID}.tif)
exten = s,n,rxfax(${FAXFILE})
exten = s,n,System(/usr/bin/mailfax ${ARG1} ${FAXFILE} ${ARG2})
exten = s,n,Hangup()
exten = T,1,Hangup()
I read that you had to put a AbsoluteTimeout in there, or it might not
hang up. My questions then are... why wouldn't it hang up without the
timeout, and what if the fax really is that large? We sometimes get
faxes over 150 pages.
Rob
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RE: [asterisk-users] Call load balancing
Never mind I found it shortly after sending this :S Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Friday, March 09, 2007 10:59 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Call load balancing Anyway, back to your question, how about your head system running an AGI that connects to the manager interface on the IVR boxes to find out how many calls each is currently processing? You could set a channel variable with the least busy host name and use that in your dial statement. If you passed the IVR host name list to the AGI, you could take a box out of service by editing and reloading your dialplan. Can you give me a link to more information about how to use the management interface? I've been having a hard time trying to track down more advanced documentation and reference material. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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
[asterisk-users] Boot order of 2 TE110P and 1 TDM400P in the same
Fedora Core 6 regards, Pablo. On Thu, Mar 08, 2007 at 05:01:15PM -0800, Jose Bertuzzi wrote: Hello Everyone, I checked with zttool that sometimes after the machine boots the order of the boards is changed like this: â Alarms Span â OK Digium Wildcard TE110P T1/E1 Card 0 â OK Digium Wildcard TE110P T1/E1 Card 1 â OK Wildcard TDM400P REV I Board 1 and sometimes: â Alarms Span â OK Wildcard TDM400P REV I Board 1â OK Digium Wildcard TE110P T1/E1 Card 1 â OK Digium Wildcard TE110P T1/E1 Card 0 What do I have to configure in order to the boards appear in the same position and the configuration work always?? Which linux distribution? -- Tzafrir Cohen icq#16849755jabber:tzafrir at jabber.org +972-50-7952406 mailto:tzafrir.cohen at xorcom.com http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir - TV dinner still cooling? Check out Tonight's Picks on Yahoo! TV.___ --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] RE: Coaching in asterisk
Wai Wu wrote: Ouch, I just have to move to 1.4. Is 1.4 stable at all under heavy load? You're more courageous than I am. -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] A New Phone Service - www.virtualphoneline.com
[EMAIL PROTECTED] wrote: Dear Asterisk Users Mailing List - Non-Commercial Discussion, I joined VirtualPhoneLine.Com service and am really enjoying the use of it. Somebody punt this jerk. -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] A New Phone Service - www.virtualphoneline.com
Anselm Martin Hoffmeister wrote: Am Dienstag, den 06.03.2007, 05:18 -0400 schrieb Chris Mason (Lists): Of course, it would be highly unlikely anyone on the list would want to report Rehan...but in case anyone does: I have been told that unsolicited commercial e-mail (I do not imply that Rehan's post fulfills the criteria, judge yourself) may be * forwarded to [EMAIL PROTECTED] * for the people there to take further measures (whatever that means). I wonder that this fact is obviously widely unknown to both citizens and US-based spammers. I further wonder wether they rate the incoming complaints by number per incident, and perhaps prefer to prioritize the cases apparently more interesting to the wide public. I am not a US citizen, so FTC probably does not care for my 2c. Worth a try, anyway. Anselm: uce.gov gets so inundated with forwarded spam that it's effectively useless. This problem is not going to be solved with anything but big changes in SMTP. -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] Another Faxing Question
In my (limited) experience with rxfax, it has issues with large faxes. I
soon gave up on rxfax and switched to hylafax (which works much better).
Check the wiki for installation instructions. (And hylafax will
correctly hangup when the fax has completed/failed/whatever.)
Wes Baehr
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rob Schall
Sent: Friday, March 09, 2007 11:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Another Faxing Question
This probably came up before, but I have a faxing question for
everyone.
I have a simple extension setup to use rxfax to receive faxes sent to
asterisk. It is:
exten = s,1,Answer()
exten = s,n,AbsoluteTimeout(300)
exten =
s,n,Set(FAXFILE=/var/spool/asterisk/fax/${ARG1}_${CALLERIDNUM}_${UNIQUEI
D}
.tif)
exten = s,n,rxfax(${FAXFILE})
exten = s,n,System(/usr/bin/mailfax ${ARG1} ${FAXFILE} ${ARG2})
exten = s,n,Hangup()
exten = T,1,Hangup()
I read that you had to put a AbsoluteTimeout in there, or it might not
hang up. My questions then are... why wouldn't it hang up without the
timeout, and what if the fax really is that large? We sometimes get
faxes over 150 pages.
Rob
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RE: [asterisk-users] When to use Echo Cancellation cards?
With hw echo cancellation you are pretty much guaranteed to not have any problems. At least with Sangoma cards. I cannot speak for the other manufacturers. I believe most of not all HWEC also does other things to help clean up the sound and maybe even add background noise etc. so the over all sound quality is improved as well. Some installations will work fine just with the included SWEC, some will not. Some will sort of kind of sometimes work ok maybe. No scientific way to determine if you will need it or not. I believe the farther away you are from the phone companies exchange the more likely you will need it. With Sangoma cards, I believe you have the option of ordering without HWEC from your supplier and if you find you need it, you can send the card to Sangoma and they will upgrade it for you or send you a HWEC version and charge you the difference. If you phone Sangoma they will be able to answer that more difinitively. -Original Message- From: Zeeshan Zakaria [mailto:[EMAIL PROTECTED] Sent: Thursday, March 08, 2007 11:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] When to use Echo Cancellation cards? In what scenarios non-Echo Cancellation cards (T1/E1/FXO) should be used ? Don't all good and professional installations need echo cancellation cards? Are there people out there with non-Echo Cancellation cards for T1 or 8 FXO ports and who really don't have any echo issues and they are running serious businesses? -- Zeeshan A Zakaria ___ --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] RE: Coaching in asterisk
I didn't know you are courageous. I upgraded to 1.4 last night. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Friday, March 09, 2007 11:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Coaching in asterisk Wai Wu wrote: Ouch, I just have to move to 1.4. Is 1.4 stable at all under heavy load? You're more courageous than I am. -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 ___ --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] RE: Coaching in asterisk
BTW. We only use Asterisk for a few functions. Everything else is done on an extenal application controlling Asterisk through AMI. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Friday, March 09, 2007 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] RE: Coaching in asterisk I didn't know you are courageous. I upgraded to 1.4 last night. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Friday, March 09, 2007 11:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Coaching in asterisk Wai Wu wrote: Ouch, I just have to move to 1.4. Is 1.4 stable at all under heavy load? You're more courageous than I am. -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 ___ --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 load balancing
I've got a system I'm putting together to handle IVR calls with *
I have one head system that terminates two PRIs. It routes the calls from
the PRIs to * boxes using IAX I'm planning on having four or five * boxes.
The * boxes run AGI scripts to process the IVR calls. Can I load balance the
routing if I have five calls each of the IVR * boxes gets two call and the
next call would go to the system that currently has the lowest number of
calls?
Another approach: what about load-balance (in terms of redundancy and
scalability) the AGI app's and just the AGIs with FastAGI? So your
IVR application can be separated from your * boxes and they (the * boxes)
dont have to ve overloaded with your AGI apps.
Your head system receive the two PRIs and in dial-plan logic you can (maybe
using RANDOM() or something more deterministic like a counter)
[just an example]:
exten s,1,Answer
exten s,n,Random(50:next)
exten s,n,AGI(agi://asterisk1/${VAR1}|${VAR2})
exten s,n,Hangup
exten s,n,AGI(agi://asterisk2/${VAR1}|${VAR2})
exten s,n,Hangup
--
Honi soit la vache qui rit.
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[asterisk-users] Which hylafax client ?
Hi, Which Hylafax client do you use. I'm after something cheap, you could use from Windows XP, as a virtual printer and that could retrieve fax numbers from an existing directory (Windows Address Book or Outlook or LDAP). Regards ___ --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] Which hylafax client ?
Olivier, For a list of your many options, see: http://www.hylafax.org/content/Desktop_Client_Software I'm partial to HylaFSP, but we sell it so can hardly be considered objective. ;-) -Darren - Original Message - From: Olivier To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, March 09, 2007 12:49 PM Subject: [asterisk-users] Which hylafax client ? Hi, Which Hylafax client do you use. I'm after something cheap, you could use from Windows XP, as a virtual printer and that could retrieve fax numbers from an existing directory (Windows Address Book or Outlook or LDAP). Regards -- ___ --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] Newbie Question
On 3/9/07, mail-lists [EMAIL PROTECTED] wrote: [test] disallow=all allow=gsm ;GSM consumes far less bandwidth than ulaw ;allow=ulaw ;allow=alaw Are you sure that the xlite phone can handle gsm?? I use it on Linux and it does. -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
[asterisk-users] AEL #include file
Hi, Does anyone know how to include a file in AEL using the #include filename syntax in .conf files? Regards, Philipp -- 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 ___ --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] When to use Echo Cancellation cards?
We're not running echo cancelling cards here. We may have 1 or 2 phone calls a month with echo, and it's primarily calls to a certain number. When asked about the echo, I explained the difference in price, and for the price difference, we can deal with the echos. For the most part, for us, either software echo cancelling seems to be working or we don't have any echo to begin with. The only time I have personally ever noticed the echo is at the beginning of a conversation, and after about 10 seconds into it, it's gone. I assume that's software echo cancelling going to work. One more bit of info. When calling the number that gives us problems, the receptionist is fine, when the call is transferred is when echo shows up. That seems to be an issue on their end. But, my point is that you don't always need hardware echo cancelling. Our faxes work inbound and outbound perfectly. Not having hardware echo cancelling, I'm sure, has a lot to do with that. ___ --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] Cdr_mysql compile question
I'm reading voip-info.org http://www.voip-info.org/wiki-Asterisk+cdr+mysql Sorry if this is a dumb question, but: It says I need mysql and mysql-devel to compile cdr_mysql, but I don't want mysql on my asterisk box I want to connect to a remote mysql server. Can I use mysqlclient and mysqlclient-devel? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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
Re: [asterisk-users] AEL #include file
Philipp Kempgen wrote: Does anyone know how to include a file in AEL using the #include filename syntax in .conf files? Seems like #include test.ael works but #include test.conf does not. Regards, Philipp -- 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 ___ --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] AEL #include file
On Fri, Mar 09, 2007 at 07:49:45PM +0100, Philipp Kempgen wrote: Hi, Does anyone know how to include a file in AEL using the #include filename syntax in .conf files? Yes, it is supported. (Technically: It is not part of the ael syntax. #include and #exec are preprocessing done before the ael parser gets to read the text.) -- 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] Cdr_mysql compile question
Nevermind, this was a dumb question :( Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Ruggles Sent: Friday, March 09, 2007 1:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Cdr_mysql compile question I'm reading voip-info.org http://www.voip-info.org/wiki-Asterisk+cdr+mysql Sorry if this is a dumb question, but: It says I need mysql and mysql-devel to compile cdr_mysql, but I don't want mysql on my asterisk box I want to connect to a remote mysql server. Can I use mysqlclient and mysqlclient-devel? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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
[asterisk-users] OS X Frequent console disconnects 1.4.1
Hi, I'm seeing the following message in the full log: WARNING[478] asterisk.c: poll returned 0: Bad file descriptor it's repeated a number of times then I'm disconnected from the running asterisk instance. What's the best way to correctly report this? ___ --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] AEL #include file
Tzafrir Cohen wrote: On Fri, Mar 09, 2007 at 07:49:45PM +0100, Philipp Kempgen wrote: Hi, Does anyone know how to include a file in AEL using the #include filename syntax in .conf files? Yes, it is supported. (Technically: It is not part of the ael syntax. #include and #exec are preprocessing done before the ael parser gets to read the text.) Is there a way to include a .conf file from within .ael? Or the other way round? BTW: Never heard of #exec. What does that do? Shell exec? Regards, Philipp -- 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 ___ --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] disable client side hangup after dialing 911
Off topic: I usually joke with students about response codes to a SIP bye request: What happens if you send a BYE and the other side responds 603 declined ? - I don't want to hangup, I want to continue talking Mother-in-laws would love that... /O ___ --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] play file and action only stop if one defined key has been pressed
Hi, I would like that user cann press 3 and then actions can be taken. Problem ist if the pressed key not 3 the user jumps to extension i and then the file will be played from start again. I would like that the play of file is only stopped if the user has pressed the key 3. What for an command can i use to make this happened? exten = i,1,GoTo(restart) exten = 3,1,NoOp(action) exten = s,n(restart),Background(file) best regards 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: Re: [asterisk-users] How to enter bridge_native_loop???
Hi, With canreinvite=yes, all the media/rtp traffic for the call typically flows directly between the two peers. So how is the code in bridge_native_loop called and when? Is it called and used for any further sip signalling and not rtp? Thanks for your prompt reply. Regards, Santosh. Hi, I am using asterisk-1.4.0. I am inquisitive of what Packet2Packet bridge (bridge_p2p_loop) does and what Native bridge (bridge_native_loop) does. I have configured my dial plans and options such that I can enter bridge_p2p_loop. However, I am unable to enter bridge_native_loop for some reason. I have the following extensions: exten = 7126,1,Dial(SIP/lin_santosh) exten = 7126,s+1,Hangup exten = 7140,1,Dial(SIP/win_test) exten = 7140,s+1,Hangup My sip.conf is as: [lin_santosh] type=friend regexten=7126 callerid=LIN Santosh 7126 host=dynamic nat=yes canreinvite=no allow=all You have set canreinvite to no, thus disabling native briding. /O ___ --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 1.2.15 chan_vpb with vpb-driver 4.0
Yifan Zhang wrote: Hi, all, I am using Asterisk 1.2.15 with an OpenLine4 card (vpb-driver 4.0). And Asterisk segfaults. Here is the output of loading chan_vpb. Very detailed because I turned on vpb verbose. any lead to solution will be appreciated. Thanks This has nothing to do with Polycom phones. Please start a new thread. Thanks! -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] IAX2, DTMF and x86_64.
William F. Acker WB2FLW +1-303-722-7209 wrote: Hi all, I'm just starting to play with 1.4. I installed 1.4.1 on an Ia32 machine, and can't find any problems. So, I decided to upgrade my home pbx. All went well until I tried using my S101 to talk to the IVR. Some times, the first one or two digits get through, but eventually a digit will get stuck, playing continuously until the call is terminated. I have confirmed this on another x86_64 machine that I connect with. Also, when I reloaded IAX2, Asterisk crashed with a message about a double linklist and an ugly trace. Unfortunately, the crash didn't make it into the logs. Any ideas? Report the bug and revert to 1.2.x on your home pbx :\ -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] AEL #include file
On Fri, 2007-03-09 at 20:42 +0100, Philipp Kempgen wrote: Tzafrir Cohen wrote: On Fri, Mar 09, 2007 at 07:49:45PM +0100, Philipp Kempgen wrote: Hi, Does anyone know how to include a file in AEL using the #include filename syntax in .conf files? Yes, it is supported. Correct. It should assume that files are in /etc/asterisk, if that's what the config file dir is. If they are somewhere, use absolute paths. (Technically: It is not part of the ael syntax. #include and #exec are preprocessing done before the ael parser gets to read the text.) Well, mostly true; the #include directives are obeyed at the lexical level of the AEL parser, which is underneath the parser. Is there a way to include a .conf file from within .ael? Or the other way round? No, there isn't. the extensions.conf format is entirely different than the AEL format, and the AEL parser will not read in extensions.conf formatted files. BTW: Never heard of #exec. What does that do? Shell exec? The #exec option is available in the extensions.conf (and all config files, for that matter). It basically will run the command provided, and the output from it had better be the config file it wants to read in. This is NOT available for AEL files (at the moment, at least). Regards, Philipp smime.p7s Description: S/MIME cryptographic 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
[asterisk-users] Recorded file processing app wanted
Does anybody have (or know of) a command line application that would: ) Eliminate pops and other random loud noises. ) Trim leading and trailing silence. ) Trim pauses exceeding x milliseconds to y milliseconds. ) Normalize what's left. I know about normalize and have figured out how to trim leading and trailing silence in sox, but I'm looking for more :) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] Polycom call parking feature and Asterisk call parking
Hi: I want to make parking calls easier for my hard-working users. Is there a way to make the Polycom call park feature work with Asterisk? In case it just works out of the box, I haven't tried it yet; but the call park feature isn't enabled on the Polycom phones by default. -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] AEL #include file
Steve Murphy wrote:
On Fri, 2007-03-09 at 20:42 +0100, Philipp Kempgen wrote:
Tzafrir Cohen wrote:
On Fri, Mar 09, 2007 at 07:49:45PM +0100, Philipp Kempgen wrote:
Hi,
Does anyone know how to include a file in AEL using the
#include filename
syntax in .conf files?
Yes, it is supported.
Correct. It should assume that files are in /etc/asterisk, if that's
what the config file dir is. If they are somewhere, use absolute paths.
Sure.
(Technically: It is not part of the ael syntax. #include and #exec are
preprocessing done before the ael parser gets to read the text.)
Well, mostly true; the #include directives are obeyed at the lexical
level of the AEL parser, which is underneath the parser.
Ok. Thanks.
Is there a way to include a .conf file from within .ael?
Or the other way round?
No, there isn't. the extensions.conf format is entirely different than
the AEL format, and the AEL parser will not read in extensions.conf
formatted files.
Ok, that's what I tried to do. Now I understand how things
are processed.
BTW: Never heard of #exec. What does that do? Shell exec?
The #exec option is available in the extensions.conf (and all config
files, for that matter). It basically will run the command provided, and
the output from it had better be the config file it wants to read in.
Found it on the bug tracker. This seems really old but it's
not well known, is it?
This is what I do:
in extensions.conf:
context voicemail
{
mailbox = {
VoiceMailMain(${user_name},s);
}
#include e-number-vmm.ael
}
in e-number-vmm.ael:
80 = jump mailbox;
So if the user wants to change the number to reach VoicemailMain
- provided they use a web interface for that - e-number-vmm.ael
can easily be parsed and adjusted accordingly by some script.
Regards,
Philipp
--
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] play file and action only stop if one defined key has been pressed
I would like that user cann press 3 and then actions can be taken. Problem ist if the pressed key not 3 the user jumps to extension i and then the file will be played from start again. I would like that the play of file is only stopped if the user has pressed the key 3. What for an command can i use to make this happened? check http://www.voip-info.org/wiki-Asterisk+cmd+Background I think the m option is what you are looking for hth ___ --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] RE: Coaching in asterisk
Wai Wu wrote: BTW. We only use Asterisk for a few functions. Everything else is done on an extenal application controlling Asterisk through AMI. It's just that a few people have reported stability problems under load in 1.4. But if you know exactly what you want and why you're upgrading... -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] RE: Coaching in asterisk
Wai Wu wrote: I didn't know you are courageous. I upgraded to 1.4 last night. People are very sensitive about their phones working. *Very* sensitive. It's hard to be courageous in the face of an angry user. Let us know how things go. -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] play file and action only stop if one defined key has been pressed
Am Friday 09 March 2007 22:27 schrieb Time Bandit:
I would like that user cann press 3 and then actions can be taken.
Problem ist if the pressed key not 3 the user jumps to extension i and
then the file will be played from start again.
I would like that the play of file is only stopped if the user has
pressed the key 3.
What for an command can i use to make this happened?
check http://www.voip-info.org/wiki-Asterisk+cmd+Background
I think the m option is what you are looking for
thanks,
I didnt see the option.
The number can be different and is stored in mySQL
exten = ${tmp_var},1,NoOp(INFO key pressed)
exten = ${tmp_var},n,GoTo(s,restart)
is not working because when Asterisk reads the dialplan ${tmp_var} is EMPTY.
Iam now using an workaround. I have done 9 different context and depends on
the key Iam using Background with option m in each context
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[asterisk-users] REALTIME and VIEW ENTIRE DIALPLAN
Is there a way to view the entire dialplan when using Realtime? I use Realtime and MySQL connector. ___ --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] AEL #include file
On Fri, 2007-03-09 at 22:21 +0100, Philipp Kempgen wrote:
Steve Murphy wrote:
On Fri, 2007-03-09 at 20:42 +0100, Philipp Kempgen wrote:
Tzafrir Cohen wrote:
On Fri, Mar 09, 2007 at 07:49:45PM +0100, Philipp Kempgen wrote:
Hi,
Does anyone know how to include a file in AEL using the
#include filename
syntax in .conf files?
Yes, it is supported.
Correct. It should assume that files are in /etc/asterisk, if that's
what the config file dir is. If they are somewhere, use absolute paths.
Sure.
(Technically: It is not part of the ael syntax. #include and #exec are
preprocessing done before the ael parser gets to read the text.)
Well, mostly true; the #include directives are obeyed at the lexical
level of the AEL parser, which is underneath the parser.
Ok. Thanks.
Is there a way to include a .conf file from within .ael?
Or the other way round?
No, there isn't. the extensions.conf format is entirely different than
the AEL format, and the AEL parser will not read in extensions.conf
formatted files.
Ok, that's what I tried to do. Now I understand how things
are processed.
BTW: Never heard of #exec. What does that do? Shell exec?
The #exec option is available in the extensions.conf (and all config
files, for that matter). It basically will run the command provided, and
the output from it had better be the config file it wants to read in.
Found it on the bug tracker. This seems really old but it's
not well known, is it?
I can't really judge how known it is; but when it's handy, it can be
REAL handy. At Digium, for instance, we keep all our config files under
SVN, and the config
files are just #exec's for svn checkouts. All the admins have to do is
commit a change to the configs, and they will be loaded the next time
asterisk is reloaded or restarted
This is what I do:
in extensions.conf:
context voicemail
{
mailbox = {
VoiceMailMain(${user_name},s);
}
#include e-number-vmm.ael
}
in e-number-vmm.ael:
80 = jump mailbox;
So if the user wants to change the number to reach VoicemailMain
- provided they use a web interface for that - e-number-vmm.ael
can easily be parsed and adjusted accordingly by some script.
Just one little mistake I hadn't pointed out earlier; the
extensions.conf would probably really be extensions.ael ! Ignoring this,
the above is perfectly valid use of #include; just remember that the
include
would only happen when asterisk loads, reloads, or AEL is reloaded.
and, hehe, you can rotate the vmm hourly/daily to keep your users on
their toes!
Regards,
Philipp
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Re: [asterisk-users] REALTIME and VIEW ENTIRE DIALPLAN
Davis Sylvester III wrote: Is there a way to view the entire dialplan when using Realtime? I use Realtime and MySQL connector. If you mean the contents of .conf-file based merged with whatever the Realtime engine is supplying, I don't think there's a way of seeing both together. But you can use the standard CLI dialplan revelation tools in conjuction with the standard MySQL table listing tools to see everything in two pieces. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] play file and action only stop if one defined key has been pressed
On Fri, 2007-03-09 at 23:01 +0100, Thomas Winter wrote:
Am Friday 09 March 2007 22:27 schrieb Time Bandit:
I would like that user cann press 3 and then actions can be taken.
Problem ist if the pressed key not 3 the user jumps to extension i and
then the file will be played from start again.
I would like that the play of file is only stopped if the user has
pressed the key 3.
What for an command can i use to make this happened?
check http://www.voip-info.org/wiki-Asterisk+cmd+Background
I think the m option is what you are looking for
thanks,
I didnt see the option.
The number can be different and is stored in mySQL
exten = ${tmp_var},1,NoOp(INFO key pressed)
exten = ${tmp_var},n,GoTo(s,restart)
Woa! can you really do that? I would have to check the code, but I have
the strong impression that you cannot use a variable in the extension
name field, they are not evaluated, nor are they really evaluatable. All
the extensions in a context are compared when looking for a match to a
target location, but
I know that goto's etc, can use a variable in a reference, but not in a
definition like this.
murf
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Re: [asterisk-users] REALTIME and VIEW ENTIRE DIALPLAN
Brian Capouch wrote: Davis Sylvester III wrote: Is there a way to view the entire dialplan when using Realtime? I use Realtime and MySQL connector. If you mean the contents of .conf-file based merged with whatever the Realtime engine is supplying, I don't think there's a way of seeing both together. But you can use the standard CLI dialplan revelation tools in conjuction with the standard MySQL table listing tools to see everything in two pieces. B. HOw do I see the mysql stuff from the CLI. I know I can do a show dialplan from the CLI to see the .conf files stuff but not aware of how to see the mysql stuff. ___ --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] REALTIME and VIEW ENTIRE DIALPLAN
Davis Sylvester III wrote: Brian Capouch wrote: Davis Sylvester III wrote: Is there a way to view the entire dialplan when using Realtime? I use Realtime and MySQL connector. If you mean the contents of .conf-file based merged with whatever the Realtime engine is supplying, I don't think there's a way of seeing both together. But you can use the standard CLI dialplan revelation tools in conjuction with the standard MySQL table listing tools to see everything in two pieces. B. HOw do I see the mysql stuff from the CLI. I know I can do a show dialplan from the CLI to see the .conf files stuff but not aware of how to see the mysql stuff. From the dialplan, you can't. The essence of Relatime (modulo the caching that it does) is that the server *doesn't* keep configuration state that can be gotten with the Realtime engine; it looks it up dynamically. In other words, the proper tool for seeing the part that lives in DB tables is the tool that comes with the DB that extracts that information from the database backend. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] AEL #include file
Thanks for the reply! Steve Murphy wrote: At Digium, for instance, we keep all our config files under SVN, and the config files are just #exec's for svn checkouts. Nice. Just one little mistake I hadn't pointed out earlier; the extensions.conf would probably really be extensions.ael ! Right. I have that in extensions.ael. Sorry for the confusion. just remember that the include would only happen when asterisk loads, reloads, or AEL is reloaded. and, hehe, you can rotate the vmm hourly/daily to keep your users on their toes! The admin is not supposed to change the mailbox extension once a day. :-) I thought about matching _. and doing a database lookup every time but that would probably be overkill. Regards, Philipp -- 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 ___ --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] spandsp, app_rxfax: apps_Makefile.patch v1.2 v1.4 = No Workie!
Hi Guys, Looked at lotsa places on the Web/archives already. Does anyone have a Makefile for Asterisk 1.4 that integrates spandsp, app_rxfax, app_txfax? This patch sure doesn't work with the Asterisk 1.4 Makefile: http://soft-switch.org/downloads/spandsp/spandsp-0.0.2pre26/asterisk-1.2.x/apps_Makefile.patch == asterisk2:/usr/src/asterisk/asterisk/asterisk-1.4.1/apps# more Makefile # # Asterisk -- A telephony toolkit for Linux. # # Makefile for PBX applications # # Copyright (C) 1999-2006, Digium, Inc. # # This program is free software, distributed under the terms of # the GNU General Public License # -include ../menuselect.makeopts ../menuselect.makedeps C_MODS:=$(filter-out $(MENUSELECT_APPS),$(patsubst %.c,%,$(wildcard app_*.c))) CC_MODS:=$(filter-out $(MENUSELECT_APPS),$(patsubst %.cc,%,$(wildcard app_*.cc))) LOADABLE_MODS:=$(C_MODS) $(CC_MODS) ifneq ($(findstring apps,$(MENUSELECT_EMBED)),) EMBEDDED_MODS:=$(LOADABLE_MODS) LOADABLE_MODS:= endif ifneq ($(findstring ODBC_STORAGE,$(MENUSELECT_OPTS_app_voicemail)),) MENUSELECT_DEPENDS_app_voicemail+=$(MENUSELECT_DEPENDS_ODBC_STORAGE) endif ifneq ($(findstring IMAP_STORAGE,$(MENUSELECT_OPTS_app_voicemail)),) MENUSELECT_DEPENDS_app_voicemail+=$(MENUSELECT_DEPENDS_IMAP_STORAGE) endif ifeq (SunOS,$(shell uname)) MENUSELECT_DEPENDS_app_chanspy+=RT RT_LIB=-lrt endif all: _all include $(ASTTOPDIR)/Makefile.moddir_rules == Does anyone have any clues for this? ___ --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] How to best manage my dial plans as the continue to grow, and grow, and grow....
Hello List - I've been slowing growing my extensions.conf file and have been wondering how everyone manages their systems. I currently have my main extensions.conf where I reference my sub extensions (for tenants or customers) files using the include statements and define my global variables. Today while watching the asterisk console I noticed a call from a voicemail user bounced into another tenants extensions file using the # key. What i'd like to accomplish is true separation for tenants on a multi-tenant system. I'd like to remove the chance of context hopping etc... How does everyone manage their systems as they continue to grow? Thanks for reading, -- -- Christopher T Aloi -- ___ --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] Which VoIP router and switch to use for medium size business
Gordon, thanks for such a detailed and full of information email. It helped me and must have helped hundreds of others on this mailing list. In my scenario, for this client whom I am working for, their main issue has always been echo. They have about 50 extensions, with 20 in the office, busy office, calls all the time, up to 5 at any given time, 10 remote extensions and other virtual extensions just for voicemail purposes. 5 IVRs and 4 queues, one VoIP line and main trunk a T1 PRI. PRI is used for all incoming and outgoing calls except for long distance calls where VoIP line is used. I am thinking of going with HWEC and also using a good QoS switch. Right now there is only one switch (don't remember the name) and it is handling all the VoIP and data traffic. Sometimes voice breaks, and it must be because of interference from data traffic. But this is not a very serious problem and one switch with QoS should be able to handle it. Am I right here? Even if someone starts using P2P software. Current router is a linksys WRT54GL - Wireless-G Broadband Router. Is it good enough if I get a good switch? Can you suggest which switch I should get. I was looking on the Internet and found switches like Adtran NetVanta which are very expensive. What do they do which makes them so expensive? And in my case, is that the type of switch which I need or is there something cheaper out there too. I am ok without PoE. ___ --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] Polycom 501 - Auto answer on one line appearance
Chris Mason (Lists) wrote:
I am using SugarCRM together with the asterisk plugin, which allows me
to click a number, SugarCRM calls my extension then places the call when
I pickup.
I would like to have that extension auto-answer. I set it up as line 3
on my phone so normal calls do not get auto-answered. However, I have
not been able to get this to work. Has anyone implented this?
This is what I put in the config file for the phone
ringtype
se.rt.enabled=1
se.rt.1.enabled=1
se.rt.1.ringer=9
se.rt.1.type=ring
se.rt.2.enabled=1
se.rt.2.ringer=10
se.rt.2.type=ring
se.rt.3.enabled=1
se.rt.3.ringer=11
se.rt.3.timeout=1000
se.rt.3.type=ring-answer
se.rt.3.name=Ring Answer
/
The phone will only answer this if the SIP header contains a Ring
Answer flag. Asterisk has to be told to send this. For example:
exten = 300,1,SIPAddHeader(Alert-Info: RANR)
I put these in my {macaddr}-phone.cfg file:
voIpProt.SIP.alertInfo.1.class=3
voIpProt.SIP.alertInfo.1.value=RANR
The ring answer function is identified by the class number.
I hope that helps you.
-Stephen-
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Re: [asterisk-users] Polycom 501 - Auto answer on one line appearance
Chris Mason (Lists) wrote: I am using SugarCRM together with the asterisk plugin, which allows me to click a number, SugarCRM calls my extension then places the call when I pickup. I would like to have that extension auto-answer. I set it up as line 3 on my phone so normal calls do not get auto-answered. However, I have not been able to get this to work. Has anyone implented this? This is what I put in the config file for the phone ringtype se.rt.enabled=1 se.rt.1.enabled=1 se.rt.1.ringer=9 se.rt.1.type=ring se.rt.2.enabled=1 se.rt.2.ringer=10 se.rt.2.type=ring se.rt.3.enabled=1 se.rt.3.ringer=11 se.rt.3.timeout=1000 se.rt.3.type=ring-answer se.rt.3.name=Ring Answer / The problem may be that the class is global; I'm not quite sure how to make the phone only auto answer the one extension... -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
[asterisk-users] DTMF issue with TDM404
Hello list, i'm sure this is not a new issue, i'm having DTMF recognition issues with TDM404. I've already tried relaxdtmf=on/off and that did not do any good. i was wondering if there is any where else in zaptel/zapata to play with and have it fine tuning. Or maybe this card is not handeling DTMF very well?? peace! ___ --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 problem after upgrading to 1.2.16
Ah! No, there isn't a chan_zap.so in /usr/lib/asterisk/modules. I installed over a previous version, however I did delete the contents of /usr/lib/asterisk/modules before compiling and installing zaptel, libpri and asterisk. What is the best way to get chan_zap.so in there? Shouldn't that get installed when I do a make install of zaptel? Cheers, Mark. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Saturday, 10 March 2007 12:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zaptel problem after upgrading to 1.2.16 Mark Davies wrote: Hi guys, I'm hoping I've made a silly mistake here, but I've been staring at the screen for the past few hours and I can't work it out. I upgraded to 1.2.16 recently, and am having problems with zaptel. The card is detected, I get a reasonable output from ztcfg -vv, and zttool shows the installed module (TDM400) with one FXS module. But when I start asterisk, I get an error saying that my IAX connection won't work in trunked mode because there's no timing interface. Zaptel doesn't show up in the output of show channeltypes. Should there be a problem with using the trunk version of zaptel, but 1.2.16 of asterisk? Are there any places that I can specifically load/enable the zaptel module? Is there a chan_zap.so in /usr/lib/asterisk/modules? Are you upgrading over an existing install of Asterisk, or did you remove the old ver first? I had trouble building Asterisk 1.2.15 with the 1.4 zaptel driver. Asterisk did not find the zaptel src and wouldn't build chan_zap.so. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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
