[asterisk-users] Max T1s in a server?
I have seen mention of limiting a single server to no more than 100 trunks. With Sangoma having an 9 T1 card that kind of blows that limit. How many T1's can we have on a single server assuming dual Xeons? Our application calls for 122 IP Phones and 120 analog ports for faxes. Its an executive suite and we need the fax ports. No way around that requirement. Thanks, Allen. ___ --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] zaptel-1.4.0-beta2 Getting it to compile on Fedora Core 6 _64bit
I keep running into the dead end that it can't find config.h in the source tree. It looks like newer kernels don't use it anymore. Someone ran into this awhile back when compiling 1.2 and it looks as though the issue was never resolved. Any ideas? Last time I tried this, I was on fedora core 5 64bit and all went well. It's not working on this newer setup Any ideas on what I can do to make it go? THANKS! Steve make linux26 make -C /lib/modules/2.6.18-1.2849.fc6/build SUBDIRS=/usr/src/zaptel-1.4.0-beta2 modules make[1]: Entering directory `/usr/src/kernels/2.6.18-1.2849.fc6-x86_64' CC [M] /usr/src/zaptel-1.4.0-beta2/pciradio.o In file included from /usr/src/zaptel-1.4.0-beta2/zaptel.h:34, from /usr/src/zaptel-1.4.0-beta2/pciradio.c:57: /usr/src/zaptel-1.4.0-beta2/zconfig.h:9:26: error: linux/config.h: No such file or directory make[2]: *** [/usr/src/zaptel-1.4.0-beta2/pciradio.o] Error 1 make[1]: *** [_module_/usr/src/zaptel-1.4.0-beta2] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.18-1.2849.fc6-x86_64' make: *** [linux26] Error 2 -- 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] mwi for voicemail not showing up for realtimeconfig.
Got it mate. thanx for that. Am using mysql for voicemail storage, unlike in the script you've written which works on mails on disk on a certain path. All I've to do is query for INBOX(new) and Old(old) voicemessages count. cheerz - Ben Scott Keagy wrote: A while back I posted a fully functional though somewhat elaborate mechanism to get MWI working with real-time voicemail and NOT using static (static kinda takes a big chunk of value away from real-time). Search the digium Asterisk User forums for my username skeagy with keyword mwi. It does not rely on the built-in sip mechanism. It's a system of scripts that are either triggered by asterisk or a cron-job every one minute to clean out a spool directory, and it uses a uses a template SIP message in a file along with sipsak. It's been working 100% flawlessly in production for 11 months. I'm sure it would work with Asterisk 1.4beta3 assuming that voicemail.conf can still trigger an external script. Regards, Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber Sent: Monday, December 04, 2006 4:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] mwi for voicemail not showing up for realtimeconfig. Since I started using 1.4 I'm also not getting MWI. I am not using realtime. MARK. Benjamin Jacob wrote: Hello ppl, Am using realtime odbc storage for voicemail, sip users/peers, static for extensions and so on. My issue is I am not getting MWI for any fones, even tho I've got rtcachefriends=yes in sip.conf WIth tcpdump, I always see the NOTIFY going as Messages-Waiting:.no Voice-Message:.0/0.(0/0) even tho there are legitimate voicemails in the INBOX path for that particular users in the db. Any ideas, wot else shud i check for? TiA. cheerz - Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip_write warning when executing Pickup of CAPI
I'm trying to pick up a ringing SIP phone (203) across the office with exten = *9,1,Pickup(783743) where 783743 is the local part of the number that our ISDN works on. I tried all of these first: exten = *9,1,Pickup(203) exten = *9,1,Pickup(SIP/203) exten = *9,1,Pickup([EMAIL PROTECTED]) and got a declined message back from my phone (snom 300), so I then switched to picking up the ringing ISDN line (it's BT ISDN2e on a pair of Eicon Diva BRI-2M cards) The Pickup(783743) works (the phone across the room stops ringing), but the calling party gets a nasty distorted noise back down the phone, and I get dozens of these messages: Dec 5 11:37:50 WARNING[26972]: chan_sip.c:2561 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 64/64) Any ideas what I'm doing wrong? Asterisk is 1.2.11 on Debian with 2.6.8-2-686 kernel. Thanks Tom ___ --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 stop Asterisk to pick up incoming PSTN signal
Gidean Chan schrieb: Hi, How to stop Asterisk to pick up incoming PSTN signal but keep the functionality to make the call out? Thanks Gidean ___ --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 Disabling incoming PSTN calls works like that in the dialplan: [from-pstn] exten = _X.,1,NoOp() Change the context to your needs. -- F. Foerster ___ --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] Signalling but no media
Hi, I am running asterisk-1.2.10 compiled from BRI-Stuff-0.3 on a Fedora Core 5 box. We were having a problem with our firewall rules that allowed signalling through but no media when the user picked up the receiver. That has been solved now by our sys-admin guys but there were calls (SIP to SIP) that would behave in a similar way with no media. Is there anything besides security settings that could cause this? Regards, Mos ___ --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 there any Asterisk controllable thermostat?
What skills are needed to write a code yourself for X10, RS-485 or RS-232. I am planning to learn some programming so I can do the stuff myself which others haven't done yet. I once knew C/C++, and other electronic stuff, but because of not using it for years, revise and update them. ___ --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] centos 4.4 + asterisk
Hello, Are there any issues with Centos 4.4 and asterisk. Thanks in advance Varun ___ --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] any possibility of Vonage Integration
1) You can connect the vonage lines to an FXO interface. I have a customer who has the linksys router/ATA connected to FXO ports of his nortel meridian PBX switch. You might try that with digium cards, FXO port of SPA-3000 or some multiport FXO gateway. 2) Vonage softphone accounts work for incoming with asterisk. Absolute forwarding, busy forwarding and multiringing to the softphone is treated as free in-network calls. Vijay Gandhi wrote: To be more elaborate, i am using 10 vonage lines in my office, can i connect them all using asterisk, or is it possible to configure those accounts on asterisk instead of the linksys boxes i am using. Regards Vijay Gandhi -Original Message- From: Vijay Gandhi [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 05, 2006 12:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] any possibility of Vonage Integration Hello, Is there any possibility of integrating plans of vonage with asterisk. Regards Vijay Gandhi ___ --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] centos 4.4 + asterisk
I have CentOS 4.4 on several boxes with Asterisk 1.2 and they run great. Have not tested conferencing yet though. Karl Hello, Are there any issues with Centos 4.4 and asterisk. Thanks in advance Varun ___ --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] Diginetwork X100P card
Hello, I got a pair of DigiNetwork X100P FXO cards. The packet has a installation with suggest to install ' voicepet ' package. Is it really required for setting up asterisk PBX ? I am slightly confused. Anybody knows why they ask to install that package ? Please guide ? Thanks Varun ___ --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] centos 4.4 + asterisk
varun wrote: Hello, Are there any issues with Centos 4.4 and asterisk. Thanks in advance Varun ___ --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 This might be fixed in 4.4 (depends on your kernel), but: quote extra, extra, read all about it, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" /quote Regards, Ove ___ --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] centos 4.4 + asterisk
I'm currently running CentOS 4.4 64-bit with Asterisk with no problems! Ove Aursand wrote: varun wrote: Hello, Are there any issues with Centos 4.4 and asterisk. Thanks in advance Varun ___ --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 This might be fixed in 4.4 (depends on your kernel), but: quote extra, extra, read all about it, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h /quote Regards, Ove ___ --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] calls not terminating
Hi, In short - Asterisk is not able to recognize that the 'other' person to whom call was made has hung up - hence the channel stays busy. In long: I've been able to configure asterisk 1.1.12 with freepbx 2.1.3 and Digium TDM400 card (4 FXO ports). I've terminated 2 PSTN lines on these ports and making outbound calls successfully using the outbound rules. However, if the 2nd party hangs the call, this is not detected and the Flash panel (FOP) shows the ZAP channel as busy. This continues until the line is dropped on the XLITE too. Ever weirder - once my PC rebooted in process of a call, so there was no way I could hang up using the PC. The other party hung up. Even then the zap channel stayed busy. The Xlite extension which made the original call appeared as busy on FOP, but was able to make and receive calls using line-2. I had to restart the server for channel to become free and FOP to reflect that. What could be the places I look at since I am using mostly original/default settings, making only dial-plan changes? Regards, Farzal -- Farzal Ali Dojki PK: 92-21-2635021-24 | US: 1-512-STAY-UBM Telecom :: Call Centre :: Security :: Computing http://www.ubm.com.pk [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] Answer a call that is not ringing on your extension
David Parcerisa wrote: Answer a call that is not ringing on your extension. I want to pick up an external call that is ringing on another extension that is not mine. Now in my old standard pbx I press 5 and I get the call. How to do this with asterisk? See /etc/asterisk/features.conf ___ --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: Loosing IAX connection between offices
Louis-David Mitterrand wrote: Short story: IAX is still crap in 1.2.13 (haven't tested 1.4), it's unreliable and perfectly good hosts will become UNREACHABLE for no apparent reason, while SIP connections keep going through. Is this with or without the qualify= option in IAX2. ___ --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] RESEND: Blind transfer # not working for forwarded or picked calls
Roger Lewau wrote: Resending this since I got no response Hello list We have a situation where calls need to be transfered to another extension. We are using # to accomplish this but we found this is only working for calls answered at the original called extension. If the call has been forwarded to another extension or if the call has been picked up by any other phone in the same pickup group the # key does not work. How can we solve this issue? Any parameters that need to be set? We are using Asterisk 1.2.13 The t and or T option to Dial must be set. ___ --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 stop Asterisk to pick up incoming PSTN signal
Hi, How to stop Asterisk to pick up incoming PSTN signal but keep the functionality to make the call out? [from-pstn] exten = s,1,Hangup ___ --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: Loosing IAX connection between offices
On Tue, Dec 05, 2006 at 08:02:35AM -0600, Eric ManxPower Wieling wrote: Louis-David Mitterrand wrote: Short story: IAX is still crap in 1.2.13 (haven't tested 1.4), it's unreliable and perfectly good hosts will become UNREACHABLE for no apparent reason, while SIP connections keep going through. Is this with or without the qualify= option in IAX2. With or without it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip_write warning when executing Pickup of CAPI
On Tue, 5 Dec 2006, Tom Fanning wrote: I'm trying to pick up a ringing SIP phone (203) across the office with exten = *9,1,Pickup(783743) where 783743 is the local part of the number that our ISDN works on. I tried all of these first: exten = *9,1,Pickup(203) exten = *9,1,Pickup(SIP/203) exten = *9,1,Pickup([EMAIL PROTECTED]) and got a declined message back from my phone (snom 300), so I then switched to picking up the ringing ISDN line (it's BT ISDN2e on a pair of Eicon Diva BRI-2M cards) The Pickup(783743) works (the phone across the room stops ringing), but the calling party gets a nasty distorted noise back down the phone, and I get dozens of these messages: Dec 5 11:37:50 WARNING[26972]: chan_sip.c:2561 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 64/64) This sounds like a bug in Asterisk. chan-capi only accepts one format (alaw or ulaw), which is configured as native format. Asterisk here sends frames of type 64 (SLINEAR), which is wrong and not accepted by chan-capi. I cannot tell why Asterisk is doing that. Armin ___ --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 there any Asterisk controllable thermostat?
On Tue, Dec 05, 2006 at 07:57:27AM -0500, Zeeshan Zakaria wrote: What skills are needed to write a code yourself for X10, RS-485 or RS-232. I am planning to learn some programming so I can do the stuff myself which others haven't done yet. I once knew C/C++, and other electronic stuff, but because of not using it for years, revise and update them. Here is some code (one of the hits in the search for linux thermostat) http://linuxgazette.net/118/chong.html -- 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] question on tx_fax install for asterisk 1.4
I downloaded these 4 files: app_rxfax.c app_txfax.c asterisk.patch spandsp-20061130.tar.gz for use with asterisk 1.4 (these are the 1.4 files) I installed spandsp, copied app_rxfax and app_txfax into /asterisk-1.4beta3/apps my question is what do I do with asterisk.patch? I tried to put it in the apps directory also and run patch asterisk.patch and it gave me an error. can't find file to patch at input line 3 Perhaps you should have used the -p or --strip option? The text leading up to this was: -- |--- build_tools/menuselect-deps.in.orig2006-09-03 15:50:32.0 +0800 |+++ build_tools/menuselect-deps.in 2006-09-03 15:50:44.0 +0800 -- Thanks for the suggestion. Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Shared Line Appearances
anyone using/experimenting with this new feature in asterisk 1.4? is anybody able to post some info how to use and what features are supported? I have general knowledge how SLA should work, ie. monitor status of another line like BLF with additional features like answer ringing call, barge into existing call on shared line and make conference call or resume call, that was put on hold on another phone sharing line as I look in sla.conf.sample, seems, that when Dial(SIP/junky) from example, this will ring both defined phones, but what other features of SLA asterisk will support (like barge, shared line status indication)? I would like to use this feature on linksys942, because this phone doesn't support BLF, so maybe SLA will help me ;-) ; define a SLA called junky [junky] trunk = SIP/10 station = SIP/15 station = SIP/16 ___ --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 there any Asterisk controllable thermostat?
I suggest you get the code I mentioned in my last message - it is c/c++ code and as is usually the case with Linux, all the source code is there. Looking at examples is a great way to learn. Doug On Tue, 5 Dec 2006, Zeeshan Zakaria wrote: What skills are needed to write a code yourself for X10, RS-485 or RS-232. I am planning to learn some programming so I can do the stuff myself which others haven't done yet. I once knew C/C++, and other electronic stuff, but because of not using it for years, revise and update them. ___ --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] question on tx_fax install for asterisk 1.4
Hi, I had MAJOR problems with this. I ended up just putting the .so files into the asterisk modules directory. That worked for me. I can send you the files I used if it's any help to you. Regards, Jim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: 05 December 2006 14:41 To: asterisk-users@lists.digium.com Subject: [asterisk-users] question on tx_fax install for asterisk 1.4 I downloaded these 4 files: app_rxfax.c app_txfax.c asterisk.patch spandsp-20061130.tar.gz for use with asterisk 1.4 (these are the 1.4 files) I installed spandsp, copied app_rxfax and app_txfax into /asterisk-1.4beta3/apps my question is what do I do with asterisk.patch? I tried to put it in the apps directory also and run patch asterisk.patch and it gave me an error. can't find file to patch at input line 3 Perhaps you should have used the -p or --strip option? The text leading up to this was: -- |--- build_tools/menuselect-deps.in.orig2006-09-03 15:50:32.0 +0800 |+++ build_tools/menuselect-deps.in 2006-09-03 15:50:44.0 +0800 -- Thanks for the suggestion. Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] centos 4.4 + asterisk
Thanks Karl. On Tue, 2006-12-05 at 08:20 -0500, [EMAIL PROTECTED] wrote: I have CentOS 4.4 on several boxes with Asterisk 1.2 and they run great. Have not tested conferencing yet though. Karl Hello, Are there any issues with Centos 4.4 and asterisk. Thanks in advance Varun ___ --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
[asterisk-users] SER/OpenSER + Asterisk + Queue
We are in the process of redesigning our single Asterisk server that handles several queues for our clients. We offer our clients hosted queueing/call center basic services. All the agents are in remote locations behind NATs using either softphones or PAP2-like devices. What we would like to accomplish is setup a SER or OpenSER (SER) server(s) in front of our Asterisk box such that all incoming and outgoing calls are handled by SER. The basic idea is to get set up for scaleability and redundancy. The goal is to be able to add additional Asterisk servers to spread our queue loads. Nothing fancy, maybe just separate clients on different boxes (not load balancing queues across multiple Asterisk boxes since that a totally different scope of project). We could then add additional SER boxes to protect our inbound and outbound SIP gateways to our SIP providers (all our calls are SIP-based - e.g. no TDM circuits). Lastly, all our agents would register against the SER server(s) instead of directly to the Asterisk boxes. Has anyone done this? Can anyone point me to some tips/documentation? Does anyone care to comment? If agents login using AgentCallBackLogin, will Asterisk know where the agents are and send the calls to them via SER? Thank you so much in advanced. - Daniel ___ --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] Issues
Dear all, sometimes very strange things happen in * server at the office where i work. E.g. sometimes a call is committed but it suddenly hangs up. Like this output shows: Executing Dial(SIP/502-9823, Zap/7/.|60|Tt) in new stack -- Called 7/. -- Zap/7-1 answered SIP/502-9823 ... -- Hungup 'Zap/7-1' This problems most happen with the fax and pos machines that are connected to a handytone 286. I've already look for these on google in some many ways, but i couldn't find nothing interesting. I'am using Asterisk CVS-v1-0-09 Thanks in advance Regards -- Arlen Nascimento ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No ID from the calling party in SIP Header
Hi... I just started working with Asterisk and found something that looks like an error, but i want to be sure, so that's why I'm asking you. When i make a call from A to B (both SIP clients), I don't see the name of the called party in the phone that initiated the call, just the dialed number. I made an ethereal trace and found out, that there is no name during the initiation in the SIP Header? But there is a Remote-Party-ID in the SIP Packet that goes from the Server to the called party...There is nothing like P-Asserted-Id in the SIP Packet that goes to the calling party. My question... Is this an error or did i forget to activate something? The configuration of the sip.conf is: [general] language=de port=5060 disallow=all allow=alaw allow=ulaw allow=GSM nat=no canreinvite=no tos=lowdelay context=default [9001] type=friend username=9001 secret=password host=dynamic callerid=Beckenbauer, Franz 9001 context=default mailbox=9001 callgroup=1 pickupgroup=1 sendrpid=yes [9002] type=friend username=9002 secret=password host=dynamic callerid=Walter, Fritz 9002 context=default mailbox=9002 callgroup=1 pickupgroup=1 sendrpid=yes cheers, Sven ___ --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] any possibility of Vonage Integration
And if you get someone over at Vonage that knows that to do you can connect without the FXO It is like FWD you have to get the KEY from Vonage for this to work. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Paul wrote: 1) You can connect the vonage lines to an FXO interface. I have a customer who has the linksys router/ATA connected to FXO ports of his nortel meridian PBX switch. You might try that with digium cards, FXO port of SPA-3000 or some multiport FXO gateway. 2) Vonage softphone accounts work for incoming with asterisk. Absolute forwarding, busy forwarding and multiringing to the softphone is treated as free in-network calls. Vijay Gandhi wrote: To be more elaborate, i am using 10 vonage lines in my office, can i connect them all using asterisk, or is it possible to configure those accounts on asterisk instead of the linksys boxes i am using. Regards Vijay Gandhi -Original Message- From: Vijay Gandhi [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 05, 2006 12:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] any possibility of Vonage Integration Hello, Is there any possibility of integrating plans of vonage with asterisk. Regards Vijay Gandhi ___ --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 Inbound (clean). Database: 0653-2, 12/04/2006 - 12/5/2006 8:25:26 AM ___ --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] installed, stumped on sip registration
Ok, I have asterisk installed and downloaded a phone from xten.com. I just want to get it connected so I can play and went into sip.conf and commented out the xlite1 section. My xten phone keeps saying Awaiting proxy login info but in there I have entered: Username: guest Password:supersecret Domain/Realm /ip of the machine running asterisk SIP Proxy: ip of the machine running asterisk defaults for the rest. What else do I need to do so that I can call the 1000 number and hear the demo?? Thanks! ___ --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 fullcontact field contains nat device private ip
I have noticed this as well. I have seen a few configs like your DUNDi setup, that use the fullcontact URI to directly contact a phone. I was always puzzled how everyone was making this work with NAT. I have not looked into it much yet, but I wonder if the new netfilter SIP conntrack/NAT extension might help overcome this issue? Regards, David On 12/3/06, JR Richardson [EMAIL PROTECTED] wrote: Hi All, Has anyone else noticed that when a sip phone sitting behind a nat registers to asterisk using realtime database, the private IP of the phone is put into the fullcontact field instead of the public contact IP. The database has the correct public IP in the ipaddr field and correct port number in the port field, which is actually what asterisk uses to to contact the device. This eliminates the ability to use the fullcontact URI to directly contact the nat'ed phone. Works great for non-nat'ed devices. Is this by purpose or an oversight the way Realtime pulls the correct contact info in the sip registration header from the device? Does anyone know how to correct this behavior? It is the same with nat=yes or nat=no. Thanks. JR -- JR Richardson Engineering for the Masses ___ --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] Realtime question
Hello all, I was wondering if anyone has had much experience with Realtime Asterisk. I like the ability to setup my extensions and voicemail boxes in MySQL, but I have a huge worry. What if MySQL crashes. I played with rtcachefriends, but can't seem to find a way to have asterisk store the extension information to ensure the phones will continue to work even if MySQL has a hiccup. Any suggestions? Thanks. Rob ___ --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] nvlinedetect
Anyone know where I can get hold of this application (for 1.4 / trunk) ? Julian ___ --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] SetCallingPres propagation
Hello, We have several regional asterisk's connected to a central one making the the PRI calls through a TE410P card. When using SetCallingPres(prohibited) on a call at the regional level, that setting it not forwarded to the central asterisk and the call is made as if no callerid had been sent: the telco substitutes the network number. Using SetCallingPres(prohibited) on the central asterisk works though: the call is received with no callerid at all. How can I suppress callerid presentation at the regional level and keep that setting when trunking the call from regional to central asterisk's? ___ --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] installed, stumped on sip registration
On 5 Dec 2006, at 16:39, blackwater dev wrote: Ok, I have asterisk installed and downloaded a phone from xten.com. I just want to get it connected so I can play and went into sip.conf and commented out the xlite1 section. My xten phone keeps saying Awaiting proxy login info but in there I have entered: Username: guest Password:supersecret Domain/Realm /ip of the machine running asterisk SIP Proxy: ip of the machine running asterisk defaults for the rest. What else do I need to do so that I can call the 1000 number and hear the demo?? From a hazy/distant memory of the xten config, I think you need: Domain/Realm: Asterisk Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] centos 4.4 + asterisk
I am not sure but i think that fix is for compiling zaptel not asterisk . Asterisk runs on centos with 0 problems :) On 05/12/06, varun [EMAIL PROTECTED] wrote: Thanks Karl. On Tue, 2006-12-05 at 08:20 -0500, [EMAIL PROTECTED] wrote: I have CentOS 4.4 on several boxes with Asterisk 1.2 and they run great. Have not tested conferencing yet though. Karl Hello, Are there any issues with Centos 4.4 and asterisk. Thanks in advance Varun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SOLVED: DB9 e1 to RJ45 pinout
Hi guys, just to let u known the pinout for the adapter : Adapter for connect the E1 telco lines on my digium card DB9 RJ45 3 1 8 2 2 4 6 5 Adapter used for connect the Hicom150 traditional PBX on my digium card (cross connection) DB9 RJ45 3 4 8 5 2 1 6 2 All connections are in 120 ohm. It works in Italy with a E1 Sagem applaiance. Thanks to all and good work. _ Da: Giordano Grandis Inviato: venerdì 24 novembre 2006 12.34 A: 'Asterisk Users Mailing List - Non-Commercial Discussion' Oggetto: DB9 e1 to RJ45 pinout Hi all, anyone known how can i create an adapter from a DB9 port of a E1 to an RJ45 plug? My telco left active the db9 port, but on my te407p card i have rj45 connection. Anyone can help me pls ? Thanks in advance -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.430 / Virus Database: 268.14.14/548 - Release Date: 23/11/2006 15.22 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.430 / Virus Database: 268.15.7/569 - Release Date: 05/12/2006 3.00 ___ --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] Realtime Error 1045
Is there anything special that needs to be done? I am trying realtime voicemail and no matter how I set it up, be it a user in mysql or through host access rights with no username/password all I get is err 1045 which is access denied. I have all the mysql stuff installed and it writes to the cdr fine. ___ --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] regcontext, NoOp extension vanishes when extension reload and doesn't come back
Hi All, I just noticed something interesting. When a sip device registers and regcontext is setup in sip.conf, a NoOp priority 1 extension is dynamically created in the dialplan within the specified regcontext. Works great. If for some reason, modification is made to the extension.conf and a reload extension is performed, those dynamically created extensions in the regcontext vanish. Now this is ok, I understand why they vanish, but the strange thing is they don't come back when the sip device registration time expires. If I set the max regiter time of the device to be 60 seconds, after 60 seconds the phone sends another registration to the server, but since the user is already cached in, the NoOp priority 1 extension does not get re-created in the regcontext. I must perform a reload chan_sip.so, wait till the new registration hits and then the NoOp priority 1 extension is created again in the regcontext. This is a problem, if anything happens to the dialplan and it has to be reloaded, we loose active registered sip devices in the regcontext, then all hell breaks loose. Has anyone else come across this and has a work around? Ultimately, I'd like to see the regcontext function ensure the NoOp priority 1 extension is re-newed each registration cycle, whatever the time parameter is set on. Thanks. JR -- JR Richardson Engineering for the Masses ___ --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] regcontext, NoOp extension vanishes when extension reload and doesn't come back
Let me guess: The context in which you have the 2 thru n priorities is the same one as you're using for regcontext right? Don't do that, bad things will happen (as you've noticed). I'd have to review the code again, but I think what you're seeing is as a result of this. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Tuesday, December 05, 2006 1:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] regcontext,NoOp extension vanishes when extension reload and doesn't come back Hi All, I just noticed something interesting. When a sip device registers and regcontext is setup in sip.conf, a NoOp priority 1 extension is dynamically created in the dialplan within the specified regcontext. Works great. If for some reason, modification is made to the extension.conf and a reload extension is performed, those dynamically created extensions in the regcontext vanish. Now this is ok, I understand why they vanish, but the strange thing is they don't come back when the sip device registration time expires. If I set the max regiter time of the device to be 60 seconds, after 60 seconds the phone sends another registration to the server, but since the user is already cached in, the NoOp priority 1 extension does not get re-created in the regcontext. I must perform a reload chan_sip.so, wait till the new registration hits and then the NoOp priority 1 extension is created again in the regcontext. This is a problem, if anything happens to the dialplan and it has to be reloaded, we loose active registered sip devices in the regcontext, then all hell breaks loose. Has anyone else come across this and has a work around? Ultimately, I'd like to see the regcontext function ensure the NoOp priority 1 extension is re-newed each registration cycle, whatever the time parameter is set on. Thanks. JR -- JR Richardson Engineering for the Masses ___ --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 The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on tx_fax install for asterisk 1.4
Hi Jerry, Where did you find the 1.4 versions of this software? I don't see anything on the official spandsp downloads site, just pre2 and pre3 releases, no 20061130.tar.gz :) Thanks, Matt G On 05/12/06, Jerry Geis [EMAIL PROTECTED] wrote: I downloaded these 4 files: app_rxfax.c app_txfax.c asterisk.patch spandsp-20061130.tar.gz for use with asterisk 1.4 (these are the 1.4 files) I installed spandsp, copied app_rxfax and app_txfax into /asterisk-1.4beta3/apps my question is what do I do with asterisk.patch? I tried to put it in the apps directory also and run patch asterisk.patch and it gave me an error. can't find file to patch at input line 3 Perhaps you should have used the -p or --strip option? The text leading up to this was: -- |--- build_tools/menuselect-deps.in.orig2006-09-03 15:50:32.0 +0800 |+++ build_tools/menuselect-deps.in 2006-09-03 15:50:44.0 +0800 -- Thanks for the suggestion. Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] regcontext, NoOp extension vanishes when extension reload and doesn't come back
On 13:59, Tue 05 Dec 06, Watkins, Bradley wrote: Let me guess: The context in which you have the 2 thru n priorities is the same one as you're using for regcontext right? Don't do that, bad things will happen (as you've noticed). I'd have to review the code again, but I think what you're seeing is as a result of this. Then how should it be done ? I'm playing with this as well and now I'm back to 0. I just had it all working on paper... -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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] Help with dial plan - two attempts at calling agent before logging agent off?
Hi List, I'm attempting to set up a queue and agents using agent call back. This is all working fine with the queue and the agents login etc However. In my dial plan I a set variable when a call is entered into the queue to identify the origin of the call, then when the agent is called I test to see if the call is from the queue. If it is, the dial plan does not go to VM if the agent does not answer, it gives BUSY and the call is returned to the queue. The call could well be passed to the same agent again from the queue, which I am okay with - BUT I only want it to try twice before logging the agent out (just in case they have gone AWOL and not logged out). The autologoff=xx in agents.conf doesn't seem to work with agentcallback. I have tried setting another variable as a counter with some logic tests to see the number of attempts to call the agent, but this is failing as the variable appears to be lost when the call goes back to the queue. Can anyone suggest an answer to this puzzle for me. Many thanks Chris -- ___ --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-1.4.0-beta2 Getting it to compile on FedoraCore 6 _64bit
Steve, It seems like you don't have the full sources on your linux box. Do you have a directory /usr/src/linux, which is a soft link to /usr/src/kernel/2.6.18.-1.2849 ? If not, or if the directory is empty means that you need to complete your sources first. I have this version of Zaptel running ok on 2.6.18-1.2239.fc5, but you can see on your errors that linux/config.h says no such directory Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Gladden Sent: Tuesday, December 05, 2006 4:21 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] zaptel-1.4.0-beta2 Getting it to compile on FedoraCore 6 _64bit I keep running into the dead end that it can't find config.h in the source tree. It looks like newer kernels don't use it anymore. Someone ran into this awhile back when compiling 1.2 and it looks as though the issue was never resolved. Any ideas? Last time I tried this, I was on fedora core 5 64bit and all went well. It's not working on this newer setup Any ideas on what I can do to make it go? THANKS! Steve make linux26 make -C /lib/modules/2.6.18-1.2849.fc6/build SUBDIRS=/usr/src/zaptel-1.4.0-beta2 modules make[1]: Entering directory `/usr/src/kernels/2.6.18-1.2849.fc6-x86_64' CC [M] /usr/src/zaptel-1.4.0-beta2/pciradio.o In file included from /usr/src/zaptel-1.4.0-beta2/zaptel.h:34, from /usr/src/zaptel-1.4.0-beta2/pciradio.c:57: /usr/src/zaptel-1.4.0-beta2/zconfig.h:9:26: error: linux/config.h: No such file or directory make[2]: *** [/usr/src/zaptel-1.4.0-beta2/pciradio.o] Error 1 make[1]: *** [_module_/usr/src/zaptel-1.4.0-beta2] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.18-1.2849.fc6-x86_64' make: *** [linux26] Error 2 -- 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 ___ --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] G.726 on Asterisk 1.4.0
I'm trying to make a new box with Asterisk 1.4.0, work with one ATA GrandStream 496 and G.726. However I modified the rtp.c as suggested for the Sipura's ATA with USE_DEPRECATED_G726=1 is not working. Someone knows about this? Thanks, Carlos Alperin ___ --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-1.4.0-beta2 Getting it to compile on FedoraCore 6 _64bit
On Tue, Dec 05, 2006 at 02:25:02PM -0500, Carlos Alperin wrote: Steve, It seems like you don't have the full sources on your linux box. Do you have a directory /usr/src/linux, which is a soft link to /usr/src/kernel/2.6.18.-1.2849 ? Actually, the source is availble at /lib/modules/2.6.18-1.2849.fc6/build No need for the link /usr/src/linux on a decent system. If not, or if the directory is empty means that you need to complete your sources first. I have this version of Zaptel running ok on 2.6.18-1.2239.fc5, but you can see on your errors that linux/config.h says no such directory Which is the error. Here is the fix: http://svn.digium.com/view/zaptel/branches/1.4/zconfig.h?r1=1471r2=1520 A quick-and-dirty fix is: sed -i '/^#include linux\/config.h/d' zconfig.h (remove that line from zconfig.h) -- 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] Attended Transfer
Dear List, I've been working with Asterisk CVS-v1-0-09 and i'm trying to enable attended transfer feature. but i just can't do it work. I've already set atxfer = * (and many other combinations) and all extensions on extensions.conf have the t and T option. But when I'm going to test, it doesn't work. Is there any other file that i have to configure in order to make it work? I've already looked at google so many times and nothing Does anybody have an idea?? Regards -- Arlen Nascimento ___ --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] installed, stumped on sip registration
Thanks Tim, I'm an asterisk newbie so am lost with the entire sip setup and conf files. On 12/5/06, Tim Panton [EMAIL PROTECTED] wrote: On 5 Dec 2006, at 16:39, blackwater dev wrote: Ok, I have asterisk installed and downloaded a phone from xten.com. I just want to get it connected so I can play and went into sip.conf and commented out the xlite1 section. My xten phone keeps saying Awaiting proxy login info but in there I have entered: Username: guest Password:supersecret Domain/Realm /ip of the machine running asterisk SIP Proxy: ip of the machine running asterisk defaults for the rest. What else do I need to do so that I can call the 1000 number and hear the demo?? From a hazy/distant memory of the xten config, I think you need: Domain/Realm: Asterisk Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] question on tx_fax install for asterisk 1.4
http://soft-switch.org/downloads/snapshots/spandsp/ ;p -- Humberto Figuera - Using Linux 2.6.17 Usuario GNU/Linux 369709 Caracas - Venezuela GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603 ___ --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-1.4.0-beta2 Getting it to compile on Fedora Core 6 _64bit
On 12/5/06, Steve Gladden [EMAIL PROTECTED] wrote: I keep running into the dead end that it can't find config.h in the source tree. I ran into this problem yesterday trying to compile ztdummy on FC6 i586. The latest Digium tarball gave me the config.h error. I was able to compile an SVN checkout of zaptel-1.4.0-beta2, but I was still unable to load ztdummy. At this point I received Unknown symbol. FYI: I did have full kernel sources available I'll try again later today... hopefully Tzafrir's quick-and-dirty fix will do the trick. David ___ --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] question on tx_fax install for asterisk 1.4
Guh! :) Thanks! Matt G On 05/12/06, Humberto Figuera [EMAIL PROTECTED] wrote: http://soft-switch.org/downloads/snapshots/spandsp/ ;p -- Humberto Figuera - Using Linux 2.6.17 Usuario GNU/Linux 369709 Caracas - Venezuela GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603 ___ --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] any possibility of Vonage Integration
On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote: And if you get someone over at Vonage that knows that to do you can connect without the FXO It is like FWD you have to get the KEY from Vonage for this to work. And more to the point there are so many VoIP providers out there, most of them cheaper, who do not require you to use a locked ATA, and thus work great with Asterisk. I number will speak IAX or SIP at your desire. Don't be fooled by the flat rates of the locked-box providers. The real rates are so low these days most people pay less paying per minute than paying a Vonage style flat rate. In addition people report if you start making really heavy usage of your Vonage flat rate so that they are losing money on you, they notice and try to stop it. $25/month will buy you close to 50 hours of urban SIP termination, it's down to half a cent in some of the big cities. Are you going to average 50 hours on the phone each month? Some people do, but most don't. (Otherwise Vonage could not even pretend it is going to make money.) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP firmware for Siemens Optipoint 410 Economy?
I have not seen anybody on the web to have found this so I thought I would check here. Anybody got this firmware? I've found firmware for the 400, but it doesn't seem to load in the 410. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: regcontext, NoOp extension vanishes when extension reload
Let me guess: The context in which you have the 2 thru n priorities is
the same one as you're using for regcontext right?
Don't do that, bad things will happen (as you've noticed).
I'd have to review the code again, but I think what you're seeing is as
a result of this.
Regards,
- Brad
No, not exactly, I have a catchall match in the regext priority 2 that
sends the call out to another context that further processes it.
regcontext is sipregistration
astreg1*CLI show dialplan sipregistration
[ Context 'sipregistration' created by 'pbx_config' ]
'53060' =1. Noop(53060)[SIP]
'53061' =1. Noop(53061)[SIP]
'53062' =1. Noop(53062)[SIP]
'53063' =1. Noop(53063)[SIP]
'53090' =1. Noop(53090)[SIP]
'53091' =1. Noop(53091)[SIP]
'53092' =1. Noop(53092)[SIP]
'i' =1. Goto(lookupdundi|${INVALID_EXTEN}|1) [pbx_config]
'_N' = 2. Goto(localcontact|${EXTEN}|1) [pbx_config]
astreg1*CLI
-= 9 extensions (9 priorities) in 1 context. =-
If I take the _N and the i exten out, and don't put
[sipregistration] in the extension.conf file, then i can reload
extensions and the NoOp extensions remain in the dial plan. Thanks
for pointing that out, I can find another solution now.
It makes sense that if [sipregistration] exist in the extension.conf
file and a reload extensions is performed, all the dynamic extensions
in that context will be removed, because they are not really there in
the first place, statically that is.
I was using the chanisavail cmd to do the local server lookups, but
was getting really sporatic results, works good in the lab but not
solid in an uncontrolled environment, live traffic. I'm wondering if
I can use a GotoIf statement to check [sipregistration] for an active
extension
Good stuff, thanks for the insight Brad.
JR
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Re: [asterisk-users] any possibility of Vonage Integration
Brad Templeton wrote: On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote: And if you get someone over at Vonage that knows that to do you can connect without the FXO It is like FWD you have to get the KEY from Vonage for this to work. And more to the point there are so many VoIP providers out there, most of them cheaper, who do not require you to use a locked ATA, and thus work great with Asterisk. I number will speak IAX or SIP at your desire. Don't be fooled by the flat rates of the locked-box providers. The real rates are so low these days most people pay less paying per minute than paying a Vonage style flat rate. In addition people report if you start making really heavy usage of your Vonage flat rate so that they are losing money on you, they notice and try to stop it. $25/month will buy you close to 50 hours of urban SIP termination, it's down to half a cent in some of the big cities. Are you going to average 50 hours on the phone each month? Some people do, but most don't. (Otherwise Vonage could not even pretend it is going to make money.) Vonage has 24/7 support. When my DID is out I don't want to wait until Monday morning. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] regcontext, NoOp extension vanishes when extension reload and doesn't come back
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michiel van Baak
Sent: Tuesday, December 05, 2006 2:09 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] regcontext,NoOp extension
vanishes when extension reload and doesn't come back
On 13:59, Tue 05 Dec 06, Watkins, Bradley wrote:
Let me guess: The context in which you have the 2 thru n
priorities
is the same one as you're using for regcontext right?
Don't do that, bad things will happen (as you've noticed).
I'd have to review the code again, but I think what you're
seeing is
as a result of this.
Then how should it be done ?
I'm playing with this as well and now I'm back to 0. I just
had it all working on paper...
You should put all of the 2 thru n priorities in a separate context and
then include the regcontext into that.
For example:
Let's say regcontext = registrations
And you have a SIP peer:
[1234]
type=peer
...
regexten=1234
You actual dialplan context should look something like:
[extensions]
exten = _1XXX,2,Dial(SIP/${EXTEN})
exten = _1XXX,3,Hangup
Include = registrations
Now, when peer 1234 registers, the registrations context will look like:
[ Context 'registrations' created by 'SIP' ]
'1234' = 1. NoOp() [SIP]
And, since the 'extensions' context includes 'registrations', any calls
that originate in 'extensions' will succeeed where they will not if 1234
is not registered.
Regards,
- Brad
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Re: [asterisk-users] any possibility of Vonage Integration
Brad Templeton, Thats a very good point. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Paul wrote: Brad Templeton wrote: On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote: And if you get someone over at Vonage that knows that to do you can connect without the FXO It is like FWD you have to get the KEY from Vonage for this to work. And more to the point there are so many VoIP providers out there, most of them cheaper, who do not require you to use a locked ATA, and thus work great with Asterisk. I number will speak IAX or SIP at your desire. Don't be fooled by the flat rates of the locked-box providers. The real rates are so low these days most people pay less paying per minute than paying a Vonage style flat rate. In addition people report if you start making really heavy usage of your Vonage flat rate so that they are losing money on you, they notice and try to stop it. $25/month will buy you close to 50 hours of urban SIP termination, it's down to half a cent in some of the big cities. Are you going to average 50 hours on the phone each month? Some people do, but most don't. (Otherwise Vonage could not even pretend it is going to make money.) Vonage has 24/7 support. When my DID is out I don't want to wait until Monday morning. ___ --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 Inbound (clean). Database: 0653-5, 12/05/2006 - 12/5/2006 3:56:28 PM ___ --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
SV: [asterisk-users] RESEND: Blind transfer # not working forforwarded or picked calls
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Eric ManxPower Wieling Skickat: den 5 december 2006 15:06 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [asterisk-users] RESEND: Blind transfer # not working forforwarded or picked calls Roger Lewau wrote: Resending this since I got no response Hello list We have a situation where calls need to be transfered to another extension. We are using # to accomplish this but we found this is only working for calls answered at the original called extension. If the call has been forwarded to another extension or if the call has been picked up by any other phone in the same pickup group the # key does not work. How can we solve this issue? Any parameters that need to be set? We are using Asterisk 1.2.13 The t and or T option to Dial must be set. --- T and t is set for the dialed extension. If you answer the dialed extension by picking up the handset you can use the # key to transfer the call. But if you pickup the call from any other extension with *8 or if you had unconditional forwarding on the dialed extension, the extension who pickup or receive the call from forwarding is not able to use the # key to further transfer the call. Kind regards Roger ___ --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] any possibility of Vonage Integration
Thanks for all the feedback on the message, if i do the vonage integration using FXo card, is there any possibility of working on G729 or GSM codec, because linksys boxes by default use G711, which consumes hell lot of B/w. Regards Vijay Gandhi -Original Message- From: Al Bochter [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 05, 2006 4:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] any possibility of Vonage Integration Brad Templeton, Thats a very good point. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Paul wrote: Brad Templeton wrote: On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote: And if you get someone over at Vonage that knows that to do you can connect without the FXO It is like FWD you have to get the KEY from Vonage for this to work. And more to the point there are so many VoIP providers out there, most of them cheaper, who do not require you to use a locked ATA, and thus work great with Asterisk. I number will speak IAX or SIP at your desire. Don't be fooled by the flat rates of the locked-box providers. The real rates are so low these days most people pay less paying per minute than paying a Vonage style flat rate. In addition people report if you start making really heavy usage of your Vonage flat rate so that they are losing money on you, they notice and try to stop it. $25/month will buy you close to 50 hours of urban SIP termination, it's down to half a cent in some of the big cities. Are you going to average 50 hours on the phone each month? Some people do, but most don't. (Otherwise Vonage could not even pretend it is going to make money.) Vonage has 24/7 support. When my DID is out I don't want to wait until Monday morning. ___ --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 Inbound (clean). Database: 0653-5, 12/05/2006 - 12/5/2006 3:56:28 PM ___ --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: regcontext, NoOp extension vanishes when extension reload
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
JR Richardson
Sent: Tuesday, December 05, 2006 3:49 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] RE: regcontext,NoOp extension
vanishes when extension reload
Let me guess: The context in which you have the 2 thru n
priorities
is the same one as you're using for regcontext right?
Don't do that, bad things will happen (as you've noticed).
I'd have to review the code again, but I think what you're
seeing is
as a result of this.
Regards,
- Brad
No, not exactly, I have a catchall match in the regext
priority 2 that sends the call out to another context that
further processes it.
The effect is the same, even though all you're doing is a Goto. The
problem stems from the fact that the context is created by
'pbx_config'. When you do an extensions reload pbx_config removes all
contexts for which it believes it is the owner and then starts from
scratch and creates all the dialplan entries in extensions.conf (OK, any
developers reading this will tell you it's more complex and it is, but
this is a close enough approximation). So when you have a context with
the same name as your regcontext defined in extensions.conf, then any
entries in that context will be removed and only the ones configured in
extensions.conf will be added back (note that below the regexten
priorities have [SIP] as the creator). It sounds like you've figured
that out on your own empirically.
For what I believe to be the 'correct' way (or at least *a* way that
won't make you pull your hair out) of working with regexten, see my
recent e-mail response Michael van Baak.
regcontext is sipregistration
astreg1*CLI show dialplan sipregistration [ Context
'sipregistration' created by 'pbx_config' ]
'53060' =1. Noop(53060)
[SIP]
'53061' =1. Noop(53061)
[SIP]
'53062' =1. Noop(53062)
[SIP]
'53063' =1. Noop(53063)
[SIP]
'53090' =1. Noop(53090)
[SIP]
'53091' =1. Noop(53091)
[SIP]
'53092' =1. Noop(53092)
[SIP]
'i' =1. Goto(lookupdundi|${INVALID_EXTEN}|1)
[pbx_config]
'_N' = 2. Goto(localcontact|${EXTEN}|1)
[pbx_config]
astreg1*CLI
-= 9 extensions (9 priorities) in 1 context. =-
If I take the _N and the i exten out, and don't put
[sipregistration] in the extension.conf file, then i can
reload extensions and the NoOp extensions remain in the dial
plan. Thanks for pointing that out, I can find another solution now.
It makes sense that if [sipregistration] exist in the
extension.conf file and a reload extensions is performed, all
the dynamic extensions in that context will be removed,
because they are not really there in the first place,
statically that is.
I was using the chanisavail cmd to do the local server
lookups, but was getting really sporatic results, works good
in the lab but not solid in an uncontrolled environment, live
traffic. I'm wondering if I can use a GotoIf statement to
check [sipregistration] for an active extension
Good stuff, thanks for the insight Brad.
No problem. I think I might have to go update the wiki (or add an
entry, I've never actually looked to see what exists) about this. It
comes up pretty often, and there definitely appears to be some
confusion.
Regards,
- Brad
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[asterisk-users] problem with asterisk - calls where both sides cannot hear each other
Hi,
I'm looking for some help with a problem in Asterisk (possibly), and I'm
confused as heck what is going on. I've updated to the latest Asterisk
version and the problem is still occur. My setup is as follows:
I've got Asterisk running on a high end Pentium-IV box running Linux
serving 5 calls, it is located in Canada. The calls come in via analog
lines through TDM400P cards to Asterisk box, which then converts it to
G729 channels to a call center in India over the Internet. Connection
between the Asterisk Server and the India call center is done via two
Cisco PIX501 devices, The call center in India is running 5 agents using
PolyCom phones, and we're using G729 to save bandwith. And yes, we
purchused 5 licenses of G729 codec.
We're using SIP and a ring all strategy, with the first agent that picks
up getting the call. The problem we're having is that about 5-10% calls
are not connecting properly. In that both sides can talk but do not hear
each other. Since we have recording in step s,5 (in the configuration
below), I can verify that it is happening. In these problematic calls,
both sides of the call talk but they cannot hear the other side at all.
I've gone through most of the documentation and spend hours on Google
search, does anyone have any idea what could be the problem? I'm willing
to provide more information if asked.
My extensions configuration is roughly the following:
[opened]
exten = s,1,SetVar(LOOP=1)
exten = s,2,Answer
exten = s,3,Wait(1)
exten = s,4,Background(open-hiq)
exten =
s,5,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/inbound/SUPPORT-${UNIQUEID})
exten = s,6,Queue(support3600)
exten = s,7,Voicemail(100|us)
exten = 1,1,Goto(opened,s,6)
exten = 500,1,Voicemail(500)
thanks,
Singer Wang
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Re: [asterisk-users] problem with asterisk - calls where both sides cannot hear each other
On Tue, Dec 05, 2006 at 04:22:35PM -0500, Singer Wang wrote:
Hi,
I'm looking for some help with a problem in Asterisk (possibly), and I'm
confused as heck what is going on. I've updated to the latest Asterisk
version and the problem is still occur. My setup is as follows:
I've got Asterisk running on a high end Pentium-IV box running Linux
serving 5 calls, it is located in Canada. The calls come in via analog
lines through TDM400P cards to Asterisk box, which then converts it to
G729 channels to a call center in India over the Internet. Connection
between the Asterisk Server and the India call center is done via two
Cisco PIX501 devices, The call center in India is running 5 agents using
PolyCom phones, and we're using G729 to save bandwith. And yes, we
purchused 5 licenses of G729 codec.
We're using SIP and a ring all strategy, with the first agent that picks
up getting the call. The problem we're having is that about 5-10% calls
are not connecting properly. In that both sides can talk but do not hear
each other. Since we have recording in step s,5 (in the configuration
below), I can verify that it is happening. In these problematic calls,
both sides of the call talk but they cannot hear the other side at all.
I've gone through most of the documentation and spend hours on Google
search, does anyone have any idea what could be the problem? I'm willing
to provide more information if asked.
My extensions configuration is roughly the following:
[opened]
exten = s,1,SetVar(LOOP=1)
exten = s,2,Answer
exten = s,3,Wait(1)
exten = s,4,Background(open-hiq)
exten =
s,5,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/inbound/SUPPORT-${UNIQUEID})
exten = s,6,Queue(support3600)
exten = s,7,Voicemail(100|us)
exten = 1,1,Goto(opened,s,6)
exten = 500,1,Voicemail(500)
thanks,
Singer Wang
Have you made sure there isn't a firewall in the way that could be blocking
your audio? You might need to punch some holes through to allow your RTP
stream.
--
Kyle Sexton
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Re: [asterisk-users] problem with asterisk - calls where both sides cannot hear each other
On Tue, 2006-12-05 at 15:32 -0600, Kyle Sexton wrote:
On Tue, Dec 05, 2006 at 04:22:35PM -0500, Singer Wang wrote:
Hi,
I'm looking for some help with a problem in Asterisk (possibly), and I'm
confused as heck what is going on. I've updated to the latest Asterisk
version and the problem is still occur. My setup is as follows:
I've got Asterisk running on a high end Pentium-IV box running Linux
serving 5 calls, it is located in Canada. The calls come in via analog
lines through TDM400P cards to Asterisk box, which then converts it to
G729 channels to a call center in India over the Internet. Connection
between the Asterisk Server and the India call center is done via two
Cisco PIX501 devices, The call center in India is running 5 agents using
PolyCom phones, and we're using G729 to save bandwith. And yes, we
purchused 5 licenses of G729 codec.
We're using SIP and a ring all strategy, with the first agent that picks
up getting the call. The problem we're having is that about 5-10% calls
are not connecting properly. In that both sides can talk but do not hear
each other. Since we have recording in step s,5 (in the configuration
below), I can verify that it is happening. In these problematic calls,
both sides of the call talk but they cannot hear the other side at all.
I've gone through most of the documentation and spend hours on Google
search, does anyone have any idea what could be the problem? I'm willing
to provide more information if asked.
My extensions configuration is roughly the following:
[opened]
exten = s,1,SetVar(LOOP=1)
exten = s,2,Answer
exten = s,3,Wait(1)
exten = s,4,Background(open-hiq)
exten =
s,5,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/inbound/SUPPORT-${UNIQUEID})
exten = s,6,Queue(support3600)
exten = s,7,Voicemail(100|us)
exten = 1,1,Goto(opened,s,6)
exten = 500,1,Voicemail(500)
thanks,
Singer Wang
Have you made sure there isn't a firewall in the way that could be blocking
your audio? You might need to punch some holes through to allow your RTP
stream.
--
Kyle Sexton
Sorry, I forgot to add one detail. Its only happening to about 5-10% of
the calls on an average day. Most of the calls goes through properly and
both sides talk and can hear each other.
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[asterisk-users] Re: regcontext, NoOp extension vanishes when extension reload, WORKING
OK this was an easy one to fix. All I had to do is RTFM. Note on the wiki:
ATTENTION: Make sure you take a look at bug report 7144
Just do what Kevin said, include the regcontext in whatever static
context you have the priority 2 extension and don't make a static
regcontext in extension.conf. Let sip module do the rest. Works
great.
Thanks Guys.
JR
On 12/5/06, JR Richardson [EMAIL PROTECTED] wrote:
Let me guess: The context in which you have the 2 thru n priorities is
the same one as you're using for regcontext right?
Don't do that, bad things will happen (as you've noticed).
I'd have to review the code again, but I think what you're seeing is as
a result of this.
Regards,
- Brad
No, not exactly, I have a catchall match in the regext priority 2 that
sends the call out to another context that further processes it.
regcontext is sipregistration
astreg1*CLI show dialplan sipregistration
[ Context 'sipregistration' created by 'pbx_config' ]
'53060' =1. Noop(53060)[SIP]
'53061' =1. Noop(53061)[SIP]
'53062' =1. Noop(53062)[SIP]
'53063' =1. Noop(53063)[SIP]
'53090' =1. Noop(53090)[SIP]
'53091' =1. Noop(53091)[SIP]
'53092' =1. Noop(53092)[SIP]
'i' =1. Goto(lookupdundi|${INVALID_EXTEN}|1) [pbx_config]
'_N' = 2. Goto(localcontact|${EXTEN}|1) [pbx_config]
astreg1*CLI
-= 9 extensions (9 priorities) in 1 context. =-
If I take the _N and the i exten out, and don't put
[sipregistration] in the extension.conf file, then i can reload
extensions and the NoOp extensions remain in the dial plan. Thanks
for pointing that out, I can find another solution now.
It makes sense that if [sipregistration] exist in the extension.conf
file and a reload extensions is performed, all the dynamic extensions
in that context will be removed, because they are not really there in
the first place, statically that is.
I was using the chanisavail cmd to do the local server lookups, but
was getting really sporatic results, works good in the lab but not
solid in an uncontrolled environment, live traffic. I'm wondering if
I can use a GotoIf statement to check [sipregistration] for an active
extension
Good stuff, thanks for the insight Brad.
JR
--
JR Richardson
Engineering for the Masses
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[asterisk-users] Install via SVN or tarball?
I'm new to Linux, as I've been using Asterisk on FreeBSD via the ports collection. My question is simple - for using the release branch of Asterisk (1.2.13 for now), should I get in the habit of using svn to retrieve the source or should I just download the tarball? Is there a best practice or a recommended installation method? Thanks in advance, Phil ___ --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] Install via SVN or tarball?
On Tue, Dec 05, 2006 at 04:50:22PM -0500, Phil Finkler wrote: I'm new to Linux, as I've been using Asterisk on FreeBSD via the ports collection. My question is simple - for using the release branch of Asterisk (1.2.13 for now), should I get in the habit of using svn to retrieve the source or should I just download the tarball? Is there a best practice or a recommended installation method? Debian, SuSE, Ubuntu and probably some other distributions (Fedora?) have their own packages of Asterisk. This is basically the equivalent of usingAsterisk from the ports collection. -- 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] problem with asterisk - calls where both sides cannot hear each other
Okay, a bit more information..
Some more information:
the non connected problem only happens to about 5-10% of the calls, the
others go through properly.. and yes, for the rest both parties can talk
and hear each other..
asterisk version:
Asterisk 1.2.13 built by root @ [hostname] on a i686 running Linux on
2006-11-14 16:53:46 UTC
We get about 50-60 calls a day with five agents.., on busy days maybe
80-100 calls
I'm setup as a call center, people call in and their calls are routed in
from analog lines through the TDM400Ps. they are connected to agents in
India via VoIP who use PolyCom IP Phones, we're using a RingAll
strategy. we have a secure IPSec tunnel between the Canada/India via
Cisco PIX501Es..
the Asterisk server has both an public (for web interface to the logs)
and private IP (10.x.x.x) and the phones have private IPs (10.x.x.x),
the traffic between them is tunneled via Cisco PIX501s.. there isn't any
NATing going on between the Asterisk and the phones..
On Tue, 2006-12-05 at 15:32 -0600, Kyle Sexton wrote:
On Tue, Dec 05, 2006 at 04:22:35PM -0500, Singer Wang wrote:
Hi,
I'm looking for some help with a problem in Asterisk (possibly), and I'm
confused as heck what is going on. I've updated to the latest Asterisk
version and the problem is still occur. My setup is as follows:
I've got Asterisk running on a high end Pentium-IV box running Linux
serving 5 calls, it is located in Canada. The calls come in via analog
lines through TDM400P cards to Asterisk box, which then converts it to
G729 channels to a call center in India over the Internet. Connection
between the Asterisk Server and the India call center is done via two
Cisco PIX501 devices, The call center in India is running 5 agents using
PolyCom phones, and we're using G729 to save bandwith. And yes, we
purchused 5 licenses of G729 codec.
We're using SIP and a ring all strategy, with the first agent that picks
up getting the call. The problem we're having is that about 5-10% calls
are not connecting properly. In that both sides can talk but do not hear
each other. Since we have recording in step s,5 (in the configuration
below), I can verify that it is happening. In these problematic calls,
both sides of the call talk but they cannot hear the other side at all.
I've gone through most of the documentation and spend hours on Google
search, does anyone have any idea what could be the problem? I'm willing
to provide more information if asked.
My extensions configuration is roughly the following:
[opened]
exten = s,1,SetVar(LOOP=1)
exten = s,2,Answer
exten = s,3,Wait(1)
exten = s,4,Background(open-hiq)
exten =
s,5,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/inbound/SUPPORT-${UNIQUEID})
exten = s,6,Queue(support3600)
exten = s,7,Voicemail(100|us)
exten = 1,1,Goto(opened,s,6)
exten = 500,1,Voicemail(500)
thanks,
Singer Wang
Have you made sure there isn't a firewall in the way that could be blocking
your audio? You might need to punch some holes through to allow your RTP
stream.
--
Kyle Sexton
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Re: [asterisk-users] any possibility of Vonage Integration
You login to your vonage account on the web and set the bandwidth saver option. That is the most you can do with a locked ATA. Vijay Gandhi wrote: Thanks for all the feedback on the message, if i do the vonage integration using FXo card, is there any possibility of working on G729 or GSM codec, because linksys boxes by default use G711, which consumes hell lot of B/w. Regards Vijay Gandhi -Original Message- *From:* Al Bochter [mailto:[EMAIL PROTECTED] *Sent:* Tuesday, December 05, 2006 4:06 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] any possibility of Vonage Integration Brad Templeton, Thats a very good point. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Paul wrote: Brad Templeton wrote: On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote: And if you get someone over at Vonage that knows that to do you can connect without the FXO It is like FWD you have to get the KEY from Vonage for this to work. And more to the point there are so many VoIP providers out there, most of them cheaper, who do not require you to use a locked ATA, and thus work great with Asterisk. I number will speak IAX or SIP at your desire. Don't be fooled by the flat rates of the locked-box providers. The real rates are so low these days most people pay less paying per minute than paying a Vonage style flat rate. In addition people report if you start making really heavy usage of your Vonage flat rate so that they are losing money on you, they notice and try to stop it. $25/month will buy you close to 50 hours of urban SIP termination, it's down to half a cent in some of the big cities. Are you going to average 50 hours on the phone each month? Some people do, but most don't. (Otherwise Vonage could not even pretend it is going to make money.) Vonage has 24/7 support. When my DID is out I don't want to wait until Monday morning. ___ --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 Inbound (clean). Database: 0653-5, 12/05/2006 - 12/5/2006 3:56:28 PM ___ --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] problem with asterisk - calls where both sides cannot hear each other
Cisco PIX501 devices, The call center in India is running 5 agents using PolyCom phones, and we're using G729 to save bandwith. And yes, we purchused 5 licenses of G729 codec. What firmware version on the Polycoms? ___ --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] Attended Transfer
Attended transfer is really four functions 1. Put the caller on Hold while you dial another number 2. Speak to the dialed number (announce the call) 3. Patch the call on hold to the other party using transfer button. 4. Disconnect (otherwise this would be a 3 party conference) How these functions work depend on what type of device the operator is using. SIP phones have this functionality ie a hold button, a transfer button and multi-line appearances. If you are using an ATA with an ordinary phone and standard dial-pad then you may be able to put a call on hold by using the * and transfer by #. But obviously one is limited to the vacant digits on the dial pad (DTMF). Note: If your analog (POTS) phone has a hold button this will not work as the hold button simply applies a resistive load to hold the loop in an off-hook status. Hope this helps... Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Dear List, I've been working with Asterisk CVS-v1-0-09 and i'm trying to enable attended transfer feature. but i just can't do it work. I've already set atxfer = * (and many other combinations) and all extensions on extensions.conf have the t and T option. But when I'm going to test, it doesn't work. Is there any other file that i have to configure in order to make it work? I've already looked at google so many times and nothing Does anybody have an idea?? Regards -- Arlen Nascimento ___ --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] any possibility of Vonage Integration
This 24/7 mantra that companies keep promoting to us is often just the ability to subject us to endless hours of their lame MOH while you wait for the one service specialist to answer the phone from Tinbuckto. My apologies if you live in Tinbukto. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada You login to your vonage account on the web and set the bandwidth saver option. That is the most you can do with a locked ATA. Vijay Gandhi wrote: Thanks for all the feedback on the message, if i do the vonage integration using FXo card, is there any possibility of working on G729 or GSM codec, because linksys boxes by default use G711, which consumes hell lot of B/w. Regards Vijay Gandhi -Original Message- *From:* Al Bochter [mailto:[EMAIL PROTECTED] *Sent:* Tuesday, December 05, 2006 4:06 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] any possibility of Vonage Integration Brad Templeton, Thats a very good point. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Paul wrote: Brad Templeton wrote: On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote: And if you get someone over at Vonage that knows that to do you can connect without the FXO It is like FWD you have to get the KEY from Vonage for this to work. And more to the point there are so many VoIP providers out there, most of them cheaper, who do not require you to use a locked ATA, and thus work great with Asterisk. I number will speak IAX or SIP at your desire. Don't be fooled by the flat rates of the locked-box providers. The real rates are so low these days most people pay less paying per minute than paying a Vonage style flat rate. In addition people report if you start making really heavy usage of your Vonage flat rate so that they are losing money on you, they notice and try to stop it. $25/month will buy you close to 50 hours of urban SIP termination, it's down to half a cent in some of the big cities. Are you going to average 50 hours on the phone each month? Some people do, but most don't. (Otherwise Vonage could not even pretend it is going to make money.) Vonage has 24/7 support. When my DID is out I don't want to wait until Monday morning. ___ --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 Inbound (clean). Database: 0653-5, 12/05/2006 - 12/5/2006 3:56:28 PM ___ --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
[asterisk-users] Meetme monitoring (once)
Has anyone found a way to monitor a meetme conference for only the first user? I find have one recording per user is pretty hard on the server performance wise... Suggestions? ___ --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] problem with asterisk - calls where both sides cannot hear each other
010100|so |3|00|Platform: Model=SoundPoint IP 301, Assembly=2345-11300-010 Rev=B 010100|so |3|00|Platform: Board=2345-11300-010 A 010100|so |3|00|Platform: MAC=0004f204e6de, IP=10.93.94.103, Subnet Mask=255.255.255.0 010100|so |3|00|Platform: BootBlock=2.5.0 (11300_010) 06-Nov-04 08:07 010100|so |3|00|Application, main: Label=BOOT, Version=3.1.3.0131 27-Jan-06 12:22 010100|so |3|00|Application, main: P/N=3150-11069-313 so we're using SoundPoint IP301s.. the Firmware from my understanding is 3.1.3.131 also, would an incorrect time setting on the phones cause this problem? On Tue, 2006-12-05 at 16:17 -0600, Eric ManxPower Wieling wrote: Cisco PIX501 devices, The call center in India is running 5 agents using PolyCom phones, and we're using G729 to save bandwith. And yes, we purchused 5 licenses of G729 codec. What firmware version on the Polycoms? ___ --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] Meetme monitoring (once)
A little more RTFM'ing and voila! Using MeetMeCount I should be able to record only the first user. http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMeCount -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Connolly Sent: Tuesday, December 05, 2006 4:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Meetme monitoring (once) Has anyone found a way to monitor a meetme conference for only the first user? I find have one recording per user is pretty hard on the server performance wise... Suggestions? ___ --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] SIP firmware for Siemens Optipoint 410 Economy?
Hi... I will send you the firmware for the 410 economy tomorrow... This firmare can only be used for the 410 economy, not for any other 410, 400 or 420 2006/12/5, Brad Templeton [EMAIL PROTECTED]: I have not seen anybody on the web to have found this so I thought I would check here. Anybody got this firmware? I've found firmware for the 400, but it doesn't seem to load in the 410. ___ --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] Question about Realtime static table
Hi All: I'd like to use Realtime Static in terms of the performance concern about dynamic realtime. Assume that I create a table: as following: CREATE TABLE `extensions_table` ( `id` int(11) NOT NULL auto_increment, `context` varchar(20) NOT NULL default '', `exten` varchar(20) NOT NULL default '', `priority` tinyint(4) NOT NULL default '0', `app` varchar(20) NOT NULL default '', `appdata` varchar(128) NOT NULL default '', PRIMARY KEY (`context`,`exten`,`priority`), KEY `id` (`id`) ) TYPE=MyISAM; Can anyone help me how to add variable names and values into the database? Thanks, Tielin ___ --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] Question about Realtime static table
This is more of a MySQL question.. But its going to look something like: ALTER TABLE `extensions_table` ADD `variable_name` type DEFAULT '0' NOT NULL ; From the specs page: http://dev.mysql.com/doc/refman/5.0/en/alter-table.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tielin Xu Sent: Tuesday, December 05, 2006 5:23 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Question about Realtime static table Hi All: I'd like to use Realtime Static in terms of the performance concern about dynamic realtime. Assume that I create a table: as following: CREATE TABLE `extensions_table` ( `id` int(11) NOT NULL auto_increment, `context` varchar(20) NOT NULL default '', `exten` varchar(20) NOT NULL default '', `priority` tinyint(4) NOT NULL default '0', `app` varchar(20) NOT NULL default '', `appdata` varchar(128) NOT NULL default '', PRIMARY KEY (`context`,`exten`,`priority`), KEY `id` (`id`) ) TYPE=MyISAM; Can anyone help me how to add variable names and values into the database? Thanks, Tielin ___ --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] Switching from FreeBSD to Linux - which distro?
Does there seem to be a popular Linux distro folks use specifically for Asterisk? I'd like to move off of FreeBSD but I'm not too familiar with Linux distros. In particular, I'm looking for a free, stable, well supported distro that has a friendly community. Any advice appreciated. Sorry for asking a question that I'm sure has been asked thousands of times. Best regards, Phil ___ --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] Help with dial plan - two attempts at calling agent before logging agent off?
snip I have tried setting another variable as a counter with some logic tests to see the number of attempts to call the agent, but this is failing as the variable appears to be lost when the call goes back to the queue. Local variables are destroyed once the call terminates. You'll have to use a global variable (yuck) or use the DB functions. Leo ___ --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] Switching from FreeBSD to Linux - which distro?
I recommend debian, been using it for years now, it was a no brainer to choose this for my asterisk deployments.. A few other people I know have used debian with asterisk with no problems either. On 12/5/06, Phil Finkler [EMAIL PROTECTED] wrote: Does there seem to be a popular Linux distro folks use specifically for Asterisk? I'd like to move off of FreeBSD but I'm not too familiar with Linux distros. In particular, I'm looking for a free, stable, well supported distro that has a friendly community. Any advice appreciated. Sorry for asking a question that I'm sure has been asked thousands of times. Best regards, Phil ___ --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] Need some examples for configuring Asterisk under Realtime static
Hi List: Can someone hlep to provide one or two examples to data entry for sip.conf under the table structure? CREATE TABLE `sip_conf` ( `id` int(11) NOT NULL auto_increment, `cat_metric` int(11) NOT NULL default '0', `var_metric` int(11) NOT NULL default '0', `commented` int(11) NOT NULL default '0', `filename` varchar(128) NOT NULL default '', `category` varchar(128) NOT NULL default 'default', `var_name` varchar(128) NOT NULL default '', `var_val` varchar(128) NOT NULL default '', PRIMARY KEY (`id`), KEY `filename_comment` (`filename`,`commented`) ) TYPE=MyISAM; Thanks, Tielin ___ --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] Switching from FreeBSD to Linux - which distro?
On Tue, 2006-12-05 at 18:47 -0500, Mike Garey wrote: I recommend debian, been using it for years now, it was a no brainer to choose this for my asterisk deployments.. A few other people I know have used debian with asterisk with no problems either. Choose Debian, is easy to maintain.. apt-get rocks ! On 12/5/06, Phil Finkler [EMAIL PROTECTED] wrote: Does there seem to be a popular Linux distro folks use specifically for Asterisk? I'd like to move off of FreeBSD but I'm not too familiar with Linux distros. In particular, I'm looking for a free, stable, well supported distro that has a friendly community. Any advice appreciated. Sorry for asking a question that I'm sure has been asked thousands of times. Best regards, Phil ___ --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 -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Switching from FreeBSD to Linux - which distro?
On Tuesday 05 December 2006 15:36, Phil Finkler wrote: Does there seem to be a popular Linux distro folks use specifically for Asterisk? I'd like to move off of FreeBSD but I'm not too familiar with Linux distros. In particular, I'm looking for a free, stable, well supported distro that has a friendly community. Any advice appreciated. Sorry for asking a question that I'm sure has been asked thousands of times. Debian is my fave, but for Asterisk I use CentOS. It's a free-of-cost clone of Red Hat Enterprise Linux, so it's very stable and reliable, and Asterisk runs great on it. Debian is good too. They have Asterisk packages, but they're generally a little bit old. Source installations work fine. Both have large, active developer and user communities. Avoid Fedora- it's too much of a moving target, and Asterisk installations are always a nightmare. Ubuntu is the darling of the Linux world, but I've had problems with Asterisk on it too. -- ~ Carla Schroder Linux geek and random computer tamer check out my Linux Cookbook! http://www.oreilly.com/catalog/linuxckbk/ best book for sysadmins and power 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
[Fwd: RE: [asterisk-users] any possibility of Vonage Integration]
I stand corrected! However you do get my point ... The bigger the company the worse it is. Having to deal with these guys is a nightmare. The company that brings me out in spots is Rogers Cable (24/7). They have this electronic air-head called Gertrude or something, (an android) who can't understand the word NO and has trouble with YES (actually like my ex-wife now that I think about it) but anyway, the point is that these companies spend millions of dollars on advertizing how much they care about you and your dog/cat/rabbit/beaver/etc. but won't spend an extra few bucks to have another person in the call center. My future policy is make a bogus call to the call center before you buy the companies product. TTFN Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada LOL.. Sorry, had to point this out: I think you meant Timbuktu... http://www.thesalmons.org/lynn/wh-timbuktu.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry.L.Coleman Sent: Tuesday, December 05, 2006 4:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] any possibility of Vonage Integration This 24/7 mantra that companies keep promoting to us is often just the ability to subject us to endless hours of their lame MOH while you wait for the one service specialist to answer the phone from Tinbuckto. My apologies if you live in Tinbukto. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada You login to your vonage account on the web and set the bandwidth saver option. That is the most you can do with a locked ATA. Vijay Gandhi wrote: Thanks for all the feedback on the message, if i do the vonage integration using FXo card, is there any possibility of working on G729 or GSM codec, because linksys boxes by default use G711, which consumes hell lot of B/w. Regards Vijay Gandhi -Original Message- *From:* Al Bochter [mailto:[EMAIL PROTECTED] *Sent:* Tuesday, December 05, 2006 4:06 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] any possibility of Vonage Integration Brad Templeton, Thats a very good point. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Paul wrote: Brad Templeton wrote: On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote: And if you get someone over at Vonage that knows that to do you can connect without the FXO It is like FWD you have to get the KEY from Vonage for this to work. And more to the point there are so many VoIP providers out there, most of them cheaper, who do not require you to use a locked ATA, and thus work great with Asterisk. I number will speak IAX or SIP at your desire. Don't be fooled by the flat rates of the locked-box providers. The real rates are so low these days most people pay less paying per minute than paying a Vonage style flat rate. In addition people report if you start making really heavy usage of your Vonage flat rate so that they are losing money on you, they notice and try to stop it. $25/month will buy you close to 50 hours of urban SIP termination, it's down to half a cent in some of the big cities. Are you going to average 50 hours on the phone each month? Some people do, but most don't. (Otherwise Vonage could not even pretend it is going to make money.) Vonage has 24/7 support. When my DID is out I don't want to wait until Monday morning. ___ --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 Inbound (clean). Database: 0653-5, 12/05/2006 - 12/5/2006 3:56:28 PM -- -- ___ --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
Re: [asterisk-users] Attended Transfer
Henry.L.Coleman wrote: Attended transfer is really four functions 1. Put the caller on Hold while you dial another number 2. Speak to the dialed number (announce the call) 3. Patch the call on hold to the other party using transfer button. 4. Disconnect (otherwise this would be a 3 party conference) How these functions work depend on what type of device the operator is using. SIP phones have this functionality ie a hold button, a transfer button and multi-line appearances. If you are using an ATA with an ordinary phone and standard dial-pad then you may be able to put a call on hold by using the * and transfer by #. But obviously one is limited to the vacant digits on the dial pad (DTMF). With an ATA you would use FLASH (aka RECALL) ___ --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] problem with asterisk - calls where both sides cannot hear each other
Singer Wang wrote: 010100|so |3|00|Platform: Model=SoundPoint IP 301, Assembly=2345-11300-010 Rev=B 010100|so |3|00|Platform: Board=2345-11300-010 A 010100|so |3|00|Platform: MAC=0004f204e6de, IP=10.93.94.103, Subnet Mask=255.255.255.0 010100|so |3|00|Platform: BootBlock=2.5.0 (11300_010) 06-Nov-04 08:07 010100|so |3|00|Application, main: Label=BOOT, Version=3.1.3.0131 27-Jan-06 12:22 010100|so |3|00|Application, main: P/N=3150-11069-313 so we're using SoundPoint IP301s.. the Firmware from my understanding is 3.1.3.131 SIP firmware would be 1.5.x, 1.6.x, or 2.0.x I recommend 1.6.7. 1.6.6 has issues, as does 2.0.x also, would an incorrect time setting on the phones cause this problem? I doubt it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto dialing: .call file vs. manager interface
Question: I'm using a .call file to make some test calls. The call file works great. When I try the same thing with the manager 'originate' action I get something weird - the originate action looks for the 's' extension in my context, regardless of what I supply as the 'extension' argument. The .call file does what I expect - it finds exten _9.,1,Noop(Looks good). The error I get in the log is as follows: Dec 5 16:44:25 VERBOSE[19670] logger.c: == Starting Zap/1-1 at autodial_start,s,1 failed so falling back to exten 's' Dec 5 16:44:25 VERBOSE[19670] logger.c: == Starting Zap/1-1 at autodial_start,s,1 still failed so falling back to context 'default' The autodial_start context looks like this: [autodial_start] exten = _9.,1,Noop(Looks good) exten = _9.,n,Goto(dialout,s,1) The dialout context just has the call handling stuff, AMD, etc. It works when the Goto works, but the Goto only seems to work when using a .call file and not the manager interface. The .call file looks like this: Channel: Zap/g0/5596221408 Callerid: 5597337550 MaxRetries: 0 RetryTime: 30 WaitTime: 30 Context: autodial_start Extension: 95596221408 Priority: 1 Account: 5898832 Has anyone experienced this issue and/or found a way around it? Thanks, MC ___ --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] Attended Transfer
Henry, according with voip-info.org, attended transfer is While on conversation with another party, you dial the atxfer key sequence. Asterisk says Transfer then gives you a dial tone, while putting the other party on hold. You dial the transferee number and talk with the transferee to introduce the call, then you can hang up and the other party will be connected with the transferee. In case the transferee does not want to answer the call, he/she simply hangs up and you will be back to your original conversation. The callee is put on hold automatically Eric, attended transfer is only possible with an ATA?? On 12/5/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Henry.L.Coleman wrote: Attended transfer is really four functions 1. Put the caller on Hold while you dial another number 2. Speak to the dialed number (announce the call) 3. Patch the call on hold to the other party using transfer button. 4. Disconnect (otherwise this would be a 3 party conference) How these functions work depend on what type of device the operator is using. SIP phones have this functionality ie a hold button, a transfer button and multi-line appearances. If you are using an ATA with an ordinary phone and standard dial-pad then you may be able to put a call on hold by using the * and transfer by #. But obviously one is limited to the vacant digits on the dial pad (DTMF). With an ATA you would use FLASH (aka RECALL) ___ --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 -- Arlen Nascimento ___ --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] Auto dialing: .call file vs. manager interface
The manager interface expects Exten NOT Extension argument header. On 12/5/06, Michael Collins [EMAIL PROTECTED] wrote: Question: I'm using a .call file to make some test calls. The call file works great. When I try the same thing with the manager 'originate' action I get something weird - the originate action looks for the 's' extension in my context, regardless of what I supply as the 'extension' argument. The .call file does what I expect - it finds exten _9.,1,Noop(Looks good). The error I get in the log is as follows: Dec 5 16:44:25 VERBOSE[19670] logger.c: == Starting Zap/1-1 at autodial_start,s,1 failed so falling back to exten 's' Dec 5 16:44:25 VERBOSE[19670] logger.c: == Starting Zap/1-1 at autodial_start,s,1 still failed so falling back to context 'default' The autodial_start context looks like this: [autodial_start] exten = _9.,1,Noop(Looks good) exten = _9.,n,Goto(dialout,s,1) The dialout context just has the call handling stuff, AMD, etc. It works when the Goto works, but the Goto only seems to work when using a .call file and not the manager interface. The .call file looks like this: Channel: Zap/g0/5596221408 Callerid: 5597337550 MaxRetries: 0 RetryTime: 30 WaitTime: 30 Context: autodial_start Extension: 95596221408 Priority: 1 Account: 5898832 Has anyone experienced this issue and/or found a way around it? Thanks, MC ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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] Attended Transfer
Arlen Nascimento wrote: Henry, according with voip-info.org, attended transfer is While on conversation with another party, you dial the atxfer key sequence. Asterisk says Transfer then gives you a dial tone, while putting the other party on hold. You dial the transferee number and talk with the transferee to introduce the call, then you can hang up and the other party will be connected with the transferee. In case the transferee does not want to answer the call, he/she simply hangs up and you will be back to your original conversation. The callee is put on hold automatically Eric, attended transfer is only possible with an ATA?? Attended transfer is supported by every decent SIP device out there. It is a basic phone feature. There are a few SIP devices out there that do NOT support attended transfer but I would not call them decent. The GS BT101 and the FREE version of X-Ten's phone are both devices that do not support attended transfer. There are a couple of reasons to want to do DTMF Transfers (configured in Asterisk via /etc/asterisk/features.conf. One reason might be that you are stuck, for some reason, with a phone that does not support attended transfer. Another reason would be if you have several different types of phones and ATAs around and do not want to make users learn different ways to do a transfer, depending on the phone the person is using at the moment. Another reason, and one I think is the most common, is that you simply don't know any better. ___ --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] TE110P Out fine / In Fail
I have just installed Asterisk wit a TE110P card. I have configured 30 channels which seems to be recognised by staff and zap show channels. I can make outbound calls with exceptional call quality but inbound (receiving) calls the caller get a message saying Your call could not be connected, please check the number and try again. Nothing is displayed in the CLI. Is this a configuration problem with Asterisk or a problem with Verizon? ___ --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: SOLVED - T1 PRI not announce this is long distance call, please add 1 for this call...
Thanks, Henry. It is very helpful for me.
I also deleted the DIAL option r in our dial out trunk which fixed the
problem.
Dial command option r: Generate a ringing tone for the calling party,
passing no audio from the called channel(s) until one answers. Without
this option, Asterisk will generate ring tones automatically where it is
appropriate to do so; however, r will force Asterisk to generate ring
tones, even if it is not appropriate. For example, if you used this
option to force ringing but the line was busy the user would hear RING
RIBEEP BEEP BEEP (thank you tzanger), which is potentially confusing
and/or unprofessional. However, the option is necessary in a couple of
places. For example, when you're dialing multiple channels, call
progress information is not consistantly passed back.
http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=5303forum=2
Cheers,
Isaac
--
Message: 11
Date: Tue, 5 Dec 2006 00:13:51 -0500 (EST)
From: Henry.L.Coleman [EMAIL PROTECTED]
Subject: Re: [asterisk-users] T1 PRI not announce this is long
distancecall, please add 1 for this call...
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain;charset=iso-8859-1
Using the PSTN in Toronto ie 416 NXX X all calls to 647 and 416
exchanges are local. 905 is an over-lapping area code, most excahnges
are
local, however Whitby (905 430 ) is Long Distance while 416 428
(Ajax) is not. You can find out which ones are long distance (from the
CRTC web site) and modify your dial plan to add the 1 to the dialed
number
or route the numbers to a DID with your friendly ITSP like Unlimitel
for
termination.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
do something like this in your extensions.conf:
exten = _NXXNXX,1,Dial(ZAP/g0/1{$EXTEN})
exten = _222NXX,1,Dial(ZAP/g0/{$EXTEN})
exten = _223NXX,1,Dial(ZAP/g0/{$EXTEN})
exten = _224NXX,1,Dial(ZAP/g0/{$EXTEN})
Where 222, 223 and 224 are local area codes.
On 12/4/06, Isaac Xiao [EMAIL PROTECTED] wrote:
Can any one help? In Toronto, we can't identify if a number is
long
distance call or not. If long distance call, we have to prefix with
1.
We
should hear a voice prompt as above to indicate that it is not a
local
call.
However, we hear the normal ring back tone (indicating the phone
had
been
connected, but actually not) when we call this long distance call
without
prefixing 1.
Here is the message shown in CLI.
Requested transfer capability: 0x00 - SPEECH
-- Called g0/9056671191
-- Zap/1-1 is proceeding passing it to SIP/9188-0e6a
-- PROGRESS with cause code 127 received
-- Zap/1-1 is making progress passing it to SIP/9188-0e6a
Thanks in advances.
Isaac
___
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[asterisk-users] Melbn Asterisk/Voip get together
Once again, it's time for the Melbourne VOIP get together. Thursday the 7th is the December meeting of the Melbourne VOIP club. We will be meeting at the usual place which is Pint on Punt, 42 Punt Road Windsor. It's near the big messy intersection of Dandenong Road, King's Way, Punt Road, Fitzroy Street and St Kilda Road. Lots of trams go by there, and Windsor train station is just up the street. Parking can be had on Peel Street. We meet in the restauranty half on the pub, up and towards the back (you'll see what I mean). We will see if we can organise a door prize. Hope to see everyone there. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any possibility of Vonage Integration
But I never said ATA. I said you call Vonage and tell Vonage that you want to B.Y.O.D. there is a KEY you need Vonage to get you and install into Asterick for Vonage service to work. Buy like Brad said there are easier ways than Vonage. I am not downing Vonage I have and still use Vonage and never had an outage with them. Yes I did install Vonage into Asterick so I know what you have to do. Just getting the right information you need from Vonage is the hard part Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Paul wrote: You login to your vonage account on the web and set the bandwidth saver option. That is the most you can do with a locked ATA. Vijay Gandhi wrote: Thanks for all the feedback on the message, if i do the vonage integration using FXo card, is there any possibility of working on G729 or GSM codec, because linksys boxes by default use G711, which consumes hell lot of B/w. Regards Vijay Gandhi -Original Message- *From:* Al Bochter [mailto:[EMAIL PROTECTED] *Sent:* Tuesday, December 05, 2006 4:06 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] any possibility of Vonage Integration Brad Templeton, Thats a very good point. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Paul wrote: Brad Templeton wrote: On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote: And if you get someone over at Vonage that knows that to do you can connect without the FXO It is like FWD you have to get the KEY from Vonage for this to work. And more to the point there are so many VoIP providers out there, most of them cheaper, who do not require you to use a locked ATA, and thus work great with Asterisk. I number will speak IAX or SIP at your desire. Don't be fooled by the flat rates of the locked-box providers. The real rates are so low these days most people pay less paying per minute than paying a Vonage style flat rate. In addition people report if you start making really heavy usage of your Vonage flat rate so that they are losing money on you, they notice and try to stop it. $25/month will buy you close to 50 hours of urban SIP termination, it's down to half a cent in some of the big cities. Are you going to average 50 hours on the phone each month? Some people do, but most don't. (Otherwise Vonage could not even pretend it is going to make money.) Vonage has 24/7 support. When my DID is out I don't want to wait until Monday morning. ___ --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 Inbound (clean). Database: 0653-5, 12/05/2006 - 12/5/2006 3:56:28 PM ___ --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 Inbound (clean). Database: 0653-5, 12/05/2006 - 12/5/2006 10:23:36 PM ___ --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] any possibility of Vonage Integration
Please hold :-) Now you will listen to MOH for 4 days :-D By the way you forgot one thing.. The person you get can't speak English. :-( Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Henry.L.Coleman wrote: This 24/7 mantra that companies keep promoting to us is often just the ability to subject us to endless hours of their lame MOH while you wait for the one service specialist to answer the phone from Tinbuckto. My apologies if you live in Tinbukto. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada You login to your vonage account on the web and set the bandwidth saver option. That is the most you can do with a locked ATA. Vijay Gandhi wrote: Thanks for all the feedback on the message, if i do the vonage integration using FXo card, is there any possibility of working on G729 or GSM codec, because linksys boxes by default use G711, which consumes hell lot of B/w. Regards Vijay Gandhi -Original Message- *From:* Al Bochter [mailto:[EMAIL PROTECTED] *Sent:* Tuesday, December 05, 2006 4:06 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] any possibility of Vonage Integration Brad Templeton, Thats a very good point. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Paul wrote: Brad Templeton wrote: On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote: And if you get someone over at Vonage that knows that to do you can connect without the FXO It is like FWD you have to get the KEY from Vonage for this to work. And more to the point there are so many VoIP providers out there, most of them cheaper, who do not require you to use a locked ATA, and thus work great with Asterisk. I number will speak IAX or SIP at your desire. Don't be fooled by the flat rates of the locked-box providers. The real rates are so low these days most people pay less paying per minute than paying a Vonage style flat rate. In addition people report if you start making really heavy usage of your Vonage flat rate so that they are losing money on you, they notice and try to stop it. $25/month will buy you close to 50 hours of urban SIP termination, it's down to half a cent in some of the big cities. Are you going to average 50 hours on the phone each month? Some people do, but most don't. (Otherwise Vonage could not even pretend it is going to make money.) Vonage has 24/7 support. When my DID is out I don't want to wait until Monday morning. ___ --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 Inbound (clean). Database: 0653-5, 12/05/2006 - 12/5/2006 3:56:28 PM ___ --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 Inbound (clean). Database: 0653-5, 12/05/2006 - 12/5/2006 10:23:43 PM ___ --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] Problem loading unicall
Hi Guys i've trying to set a mfcr2 systembut i can't find the working combination between -asterisk version -spandsp version -unicall version i can compile but loading the pbx shows [chan_unicall.so]Dec 5 23:04:42 WARNING[27865]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_unicall.so: undefined symbol: dtmf_put Dec 5 23:04:42 WARNING[27865]: loader.c:554 load_modules: Loading module chan_unicall.so failed! Any idea or the holy combination will be more than welcome Thanks Guys BR -- Yelson E Vivas C MPC (571) 6500-800 ___ --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] Rejecting a Call
All, Is there a way of rejecting a call using SIP in the Asterisk Dialplan? Essentially, I want to look at the called number and if it matches something I don't like I want to send back a SIP response which will not cause the other end to 'hunt'. The response codes that will achieve this are: 401 Unauthorized 403 Forbidden Is there a way of getting Asterisk to send back such a response in the dial plan without answering a call? Cheers, Ray ___ --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] TE110P Out fine / In Fail
30 Channels on Verizon? Is this in the US? T1 (24 channels) or E1(30 channels)? Are you dialing from the top (g1) of the group or bottom (G1)? On 12/5/06, Klaverstyn, David C [EMAIL PROTECTED] wrote: I have just installed Asterisk wit a TE110P card. I have configured 30 channels which seems to be recognised by staff and zap show channels. I can make outbound calls with exceptional call quality but inbound (receiving) calls the caller get a message saying Your call could not be connected, please check the number and try again. Nothing is displayed in the CLI. Is this a configuration problem with Asterisk or a problem with Verizon? ___ --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] centos 4.4 + asterisk
That kernel-devel fix is just for ZAPTEL. The bug has been solved in 4.4 On 12/5/06, Vicky [EMAIL PROTECTED] wrote: I am not sure but i think that fix is for compiling zaptel not asterisk . Asterisk runs on centos with 0 problems :) On 05/12/06, varun [EMAIL PROTECTED] wrote: Thanks Karl. On Tue, 2006-12-05 at 08:20 -0500, [EMAIL PROTECTED] wrote: I have CentOS 4.4 on several boxes with Asterisk 1.2 and they run great. Have not tested conferencing yet though. Karl Hello, Are there any issues with Centos 4.4 and asterisk. Thanks in advance Varun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
