Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
5 maj 2008 kl. 19.58 skrev Tilghman Lesher: On Monday 05 May 2008 11:24, Johansson Olle E wrote: 5 maj 2008 kl. 17.51 skrev Tilghman Lesher: On Monday 05 May 2008 09:45, Johansson Olle E wrote: Another issue that we need to fix with the MYSQL driver is that we're lacking a connection pool. Everything seems to be handled over one connection to Mysql, which causes issues. That's not true. The MYSQL app generally uses multiple connections, one for each channel. The only way one might use only a single connection is by using a global variable to store a single connection id, but that method is not documented anywhere, AFAIK. You talk about the Mysql APP, but is this the case with the Realtime driver as well? No, the native Realtime driver uses a single connection. The ODBC Realtime driver generally uses a single connection but can be configured to use a separate connection for each query. So, we're back to where we started. A developer that can help us with a connection pool or a separate connection for each query would be a Nice Thing (TM). /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Asterisk as root
In general, if your asterisk is accesible from the internet its much better to have it run as a non-root process. (My opinion is that this should be the default out-of-the-makefile ;) asterisk behaviour) This is the norm for more of the servers/services running on a linux system, and can act as a safety-net when things go bad Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Sent: Tuesday, May 06, 2008 3:00 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Running Asterisk as root Hi all, I have seen discussions on this earlier on, but just want to hear some quick thoughts. I am running v1.6 of Asterisk on my Ubuntu installation, I did make config to make it run at boot. Since I've got a firewall and don't have any other servers running I am not worried. I have been htinking about running Asterisk as a seperat user, but haven't done that yet. Everything is working fine. What do you think? Thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Asterisk as root
I totally agree. Someone filed a bugreport for this? Also asterisk init script should be installed by default too. I am going to give Cesar's instructions a try (sans removing /bin/sh) and hope it works! On Tue, May 6, 2008 at 3:24 AM, Stelios Koroneos [EMAIL PROTECTED] wrote: In general, if your asterisk is accesible from the internet its much better to have it run as a non-root process. (My opinion is that this should be the default out-of-the-makefile ;) asterisk behaviour) This is the norm for more of the servers/services running on a linux system, and can act as a safety-net when things go bad Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Sent: Tuesday, May 06, 2008 3:00 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Running Asterisk as root Hi all, I have seen discussions on this earlier on, but just want to hear some quick thoughts. I am running v1.6 of Asterisk on my Ubuntu installation, I did make config to make it run at boot. Since I've got a firewall and don't have any other servers running I am not worried. I have been htinking about running Asterisk as a seperat user, but haven't done that yet. Everything is working fine. What do you think? Thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Asterisk as root
Christian wrote: Hi all, I have seen discussions on this earlier on, but just want to hear some quick thoughts. I am running v1.6 of Asterisk on my Ubuntu installation, I did make config to make it run at boot. Since I've got a firewall and don't have any other servers running I am not worried. I have been htinking about running Asterisk as a seperat user, but haven't done that yet. Everything is working fine. What do you think? Thanks, Christian I'd never run a server app as root. It is just asking for trouble IMHO. When I built asterisk on my little custom linux server I documented the process of setting up as a non-privileged process here. Most of the information originally came from the voip-info.org site: http://www.theopensourcerer.com/2007/10/30/untangle-asterisk-pbx-and-file-server-all-in-one-part-7/ Hope this helps. Al -- The way out is open! http://www.theopensourcerer.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall - How to automatically block collect calls
Hi! Thank you for your answer. However I would like to know, if there is any other possibility of making this work, with older unicall versions. The reason I ask, is because it's not easy for me to upgrade the asterisk and unicall version on my machine, since it is a production machine, and I would consider a software upgrade really only if no other solution is possible. Thanks in advance for your answer. Best regards, Óscar Patrício Moises Silva escreveu: The latest version of the driver included in http://www.moythreads.com/astunicall/ comes with a change that will set the variable UC_CATEGORY in your dialplan, Brasil has a special category for those calls, don't remember the name that will show up, but you can make a couple of tests and then drop any call with that specific category. Moy On Mon, May 5, 2008 at 9:14 AM, Oscar Patricio [EMAIL PROTECTED] wrote: Hi! I am using asterisk with unicall in brasil. Everything was working fine, but now we want to set up a way to automatically drop collect calls, because we have an IVR answering all calls automatically! Can you tell me, what I have to configure to block collect calls in the asterisk? Thank you! Best regards, Óscar Patrício ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Predictive dialer - which one would you recommend?
Hi guys, I would like to ask you, if any of you has any experiences with the predictive dialers available for Asterisk? Are open source predictive dialers such as VICIDIAL Dialer any good? Which one would you recommend for a ca. 45 seat call center where most of the agents work on both inbound/outbound and are already using their own CTI software (so the predictive dialer software will be an appendix to the existing system and should be integrated with 3rd party software reasonably easy). Thanks, Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Predictive dialer - which one would you recommend?
On 5/6/08, Asterisk [EMAIL PROTECTED] wrote: Hi guys, I would like to ask you, if any of you has any experiences with the predictive dialers available for Asterisk? Are open source predictive dialers such as VICIDIAL Dialer any good? Which one would you recommend for a ca. 45 seat call center where most of the agents work on both inbound/outbound and are already using their own CTI software (so the predictive dialer software will be an appendix to the existing system and should be integrated with 3rd party software reasonably easy). Thanks, Alex I think that VICIDIAL is pretty good(full disclosure, I wrote it). It is currently in use at over 700 companies in over 70 countries around the world and is available in 9 languages. 45 seats blended are no problem for VICIDIAL, we even have some installtions running that are over 300 seats. As for integration with 3rd party software, we have integrated with many different kinds of web-based and client/server applications for our clients. Could you explain a little more exactly what functions that you would want the call center software to perform, and what functions you want your existing application to perform? Thanks, MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
On Tue, May 6, 2008 at 7:38 AM, Benoit Plessis [EMAIL PROTECTED] wrote: Hi, I'm wondering what version of asterisk people use in production environnement ? on which distribution ? And what is your setup like ? We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 and it's quite unstable. We have ~30 IAX2 SoftPhones and encounter some Avoiding IAX destroy deadlock and now that we have added a Queue, it's worse than ever. The queue goes stuck quite often (agent are stuck in 'In use' state and if they logoff they can't log-in till an asterisk restart). regards I am personally a proponent of Asterisk 1.2.X as I see more and more fatal bugs in the 1.4.X code come up on the lists as well as IAX2 bugs. I constantly hear Asterisk 1.4.whatever is much better, but the bugs coming out are not just unexpected behavior that one could live with, they are segfaults, system crashes, modules not getting installed (Zaptel). I use SIP since I have seen quite a few issues with IAX2 that were solved by simply switching to SIP. The above two yield solid systems under heavy load for me. OS is not so important I do not believe. I have some running FC8 and more running CentOS, both rock solid. I think the general consensus on OS is use what you are most familiar with. While these may not be popular opinions, I still ask, what does SwitchVox use? What do some of the guys around here that setup large systems use? Is ABE even using 1.4 yet? All I see in the ABE release notes is 1.2 although I have heard that ABE should be running 1.4 Very Soon many many moons ago http://www.digium.com/en/docs/ABE/README . So either Digium doesn't trust 1.4 enough to use it for ABE or the README is out of date. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Asterisk as root
On 2008-05-06 at 03:46 Tzafrir Cohen wrote: On Mon, May 05, 2008 at 07:18:08PM -0500, Cesar Benjamin Garcia Martinez wrote: Move to root: sudo -s type your passwd and as root: Edit the file /etc/init.d/asterisk And uncommet the two lines than sasys something like AST_USER=asterisk AST_GROUP=asterisk You need to create the user asterisk on your system. And create another symlink sh to bash: cd /bin rm -f sh ln -s bash sh Why is that? Debian / Ubuntu policy is that a script that is not posix sh should use /bin/bash. Any script of Asterisk does not fit the policy and has not bit shot^Wfixed yet? The fix is to edit the ofending script: #!/bin/sh - #!/bin/bash Edit your /etc/asterisk/asterisk.conf and replace the line: astrundir = /var/run With: astrundir = /var/lib/asterisk/var/run /var/run/asterisk Everything under /var/run is deleted at boot with Ubuntu, so the init.d script should recreate that directory and give it proper permissions if it does not exist. (or use the one from the Asterisk package) Create that folder: mkdir -p /var/lib/asterisk/var/run /var/run/asterisk, as mentioned above. and it should be created in the init.d script . and, chown to asterisk:asterisk the folders: /var/lib/asterisk/ /usr/lib/asterisk/ No real need for /usr/lib/asterisk to be owned by Asterisk. It is read-only. /usr is read-only, as you recall. /var/log/asterisk/ chown -Rv asterisk:asterisk /var/lib/asterisk/ # chown -Rv asterisk:asterisk /usr/lib/asterisk/ chown -Rv asterisk:asterisk /var/log/asterisk/ that's all Btw... delete the symlink sh - dash into /bin NOT Start daemon /etc/init.d/asterisk start -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir So what instructions are correct? I don't want to do anything that might not work. Many thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX issues with 1.4.19.1
On Mon, 2008-05-05 at 16:36 -1000, Julian Yap wrote: That was a bug in the release. From the 1.4.20-rc1 Changelog: 2008-04-30 16:30 + [r114891] Russell Bryant [EMAIL PROTECTED] So basically, r114891 was a fix to AST-2008-006? So if you applied the patch for AST-2008-006 you now really need this new fix (r114891) to regain the stability that chan_iax2.c had before the AST-2008-006 patch? b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX issues with 1.4.19.1
Yes. Julian Brian J. Murrell wrote: On Mon, 2008-05-05 at 16:36 -1000, Julian Yap wrote: That was a bug in the release. From the 1.4.20-rc1 Changelog: 2008-04-30 16:30 + [r114891] Russell Bryant [EMAIL PROTECTED] So basically, r114891 was a fix to AST-2008-006? So if you applied the patch for AST-2008-006 you now really need this new fix (r114891) to regain the stability that chan_iax2.c had before the AST-2008-006 patch? b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX issues with 1.4.19.1
On Tue, 2008-05-06 at 13:23 +0100, Julian Lyndon-Smith wrote: Yes. Hrm. For those of us that are following along the AST-* train, patching as per the AST-* release notices, as a matter of process, wouldn't it have been good to republish AST-2008-006 and include this fix along with the original patch? IOW, IMHO, it should be standard practice that when you release a fix to a patch in an AST-* security release, that you re-publish the security notice complete with the original security fix and the security fix fix. Some of us don't have the bandwidth to keep upgrading to release-of-the-week or to watch every commit looking for both the security fixes and the fixes to the security fixes. Your consideration of this process enhancement would be greatly appreciated. Thanx, b. /me goes off to patch his * once again. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
Vinícius Fontes wrote: There were some really unstable Asterisk releases in the 1.4 branch. I personally use 1.4.13 or 1.4.15 in production. Every single time I tried 1.4.16 or higher I had problems. Att Vinícius Fontes Desenvolvimento Canall Tecnologia em Comunicações Ltda. - Steve Totaro [EMAIL PROTECTED] escreveu: On Tue, May 6, 2008 at 7:38 AM, Benoit Plessis [EMAIL PROTECTED] wrote: Hi, I'm wondering what version of asterisk people use in production environnement ? on which distribution ? And what is your setup like ? We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 and it's quite unstable. We have ~30 IAX2 SoftPhones and encounter some Avoiding IAX destroy deadlock and now that we have added a Queue, it's worse than ever. The queue goes stuck quite often (agent are stuck in 'In use' state and if they logoff they can't log-in till an asterisk restart). regards I am personally a proponent of Asterisk 1.2.X as I see more and more fatal bugs in the 1.4.X code come up on the lists as well as IAX2 bugs. I constantly hear Asterisk 1.4.whatever is much better, but the bugs coming out are not just unexpected behavior that one could live with, they are segfaults, system crashes, modules not getting installed (Zaptel). I use SIP since I have seen quite a few issues with IAX2 that were solved by simply switching to SIP. The above two yield solid systems under heavy load for me. OS is not so important I do not believe. I have some running FC8 and more running CentOS, both rock solid. I think the general consensus on OS is use what you are most familiar with. While these may not be popular opinions, I still ask, what does SwitchVox use? What do some of the guys around here that setup large systems use? Is ABE even using 1.4 yet? All I see in the ABE release notes is 1.2 although I have heard that ABE should be running 1.4 Very Soon many many moons ago http://www.digium.com/en/docs/ABE/README . So either Digium doesn't trust 1.4 enough to use it for ABE or the README is out of date. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I use 1.4.18 with no problems. We have quite a few users(125 total between branches), but the call volume at the most has been around 15 active calls at a time. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
lordfuknowsyou a écrit : Vinícius Fontes wrote: I use 1.4.18 with no problems. We have quite a few users(125 total between branches), but the call volume at the most has been around 15 active calls at a time. Any IAX2 phone or mostly SIP ? Do you use Call Queues ? We have less user than that, less concurrent call but quite a few crash/deadlocks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote: 5 maj 2008 kl. 19.58 skrev Tilghman Lesher: On Monday 05 May 2008 11:24, Johansson Olle E wrote: 5 maj 2008 kl. 17.51 skrev Tilghman Lesher: On Monday 05 May 2008 09:45, Johansson Olle E wrote: Another issue that we need to fix with the MYSQL driver is that we're lacking a connection pool. Everything seems to be handled over one connection to Mysql, which causes issues. That's not true. The MYSQL app generally uses multiple connections, one for each channel. The only way one might use only a single connection is by using a global variable to store a single connection id, but that method is not documented anywhere, AFAIK. You talk about the Mysql APP, but is this the case with the Realtime driver as well? No, the native Realtime driver uses a single connection. The ODBC Realtime driver generally uses a single connection but can be configured to use a separate connection for each query. So, we're back to where we started. A developer that can help us with a connection pool or a separate connection for each query would be a Nice Thing (TM). What issues are you specifically seeing that merit using multiple connections? -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote: lordfuknowsyou a écrit : Vinícius Fontes wrote: I use 1.4.18 with no problems. We have quite a few users(125 total between branches), but the call volume at the most has been around 15 active calls at a time. Any IAX2 phone or mostly SIP ? Do you use Call Queues ? We have less user than that, less concurrent call but quite a few crash/deadlocks Have you reported these issues on the bugtracker? -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
On Tue, 2008-05-06 at 07:58 -0400, Steve Totaro wrote: [snip] While these may not be popular opinions, I still ask, what does SwitchVox use? Not sure what Asterisk version they use but I saw (iirc) a presentation on their website that they run switchvox on top of Fedora Core 6. FC6 has been end-of-line for a long, long time... Regards, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] using cell phone as an FXO port
Hi all, I want to use a cell phone as my FXO line to Asterisk Box ,did anyone try this and configured it and how to physically connect it to Asterisk server?___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using cell phone as an FXO port
On Mon, May 5, 2008 at 9:29 PM, gmail [EMAIL PROTECTED] wrote: Hi all, I want to use a cell phone as my FXO line to Asterisk Box ,did anyone try this and configured it and how to physically connect it to Asterisk server? Check out chan_mobile. Super cool. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
On Tuesday 06 May 2008 06:58:39 Steve Totaro wrote: While these may not be popular opinions, I still ask, what does SwitchVox use? What do some of the guys around here that setup large systems use? Is ABE even using 1.4 yet? Yes, ABE version C (in release for several months) is using the 1.4 codebase. All I see in the ABE release notes is 1.2 although I have heard that ABE should be running 1.4 Very Soon many many moons ago http://www.digium.com/en/docs/ABE/README . So either Digium doesn't trust 1.4 enough to use it for ABE or the README is out of date. The first clue should be that the copyright listed in that file is from 2006. Yes, it's very much out of date. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX issues with 1.4.19.1
On Tuesday 06 May 2008 07:35:14 Brian J. Murrell wrote: On Tue, 2008-05-06 at 13:23 +0100, Julian Lyndon-Smith wrote: Yes. Hrm. For those of us that are following along the AST-* train, patching as per the AST-* release notices, as a matter of process, wouldn't it have been good to republish AST-2008-006 and include this fix along with the original patch? IOW, IMHO, it should be standard practice that when you release a fix to a patch in an AST-* security release, that you re-publish the security notice complete with the original security fix and the security fix fix. It's not actually a fix to the security fix. The security fix simply highlighted an issue which was already present in Asterisk. You would have seen this same issue prior to the security fix simply by running 'iax2 show channels', as that command runs through every single slot in the channel array, looking for active channels. The security issue fix simply made that slow search process (already there, nothing new) painfully obvious. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
Mostly SIP, some of my clients have queues and everything is working fine by now. Att Vinícius Fontes Desenvolvimento Canall Tecnologia em Comunicações Ltda. - Benoit Plessis [EMAIL PROTECTED] escreveu: lordfuknowsyou a écrit : Vinícius Fontes wrote: I use 1.4.18 with no problems. We have quite a few users(125 total between branches), but the call volume at the most has been around 15 active calls at a time. Any IAX2 phone or mostly SIP ? Do you use Call Queues ? We have less user than that, less concurrent call but quite a few crash/deadlocks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan, Extensions, etc. Worksheet
I still have not had time to dig and find what I have but there are several worksheets ranging from sizing or initial customer questionnaires. This will give you an idea of what kind of hardware you will need to purchase to put together a (hardware) quote. Another worksheet goes over features, options, dialplan, telco, ISP, network and some other things. This enables you to come up with a Scope of Work and give you an idea of what to charge for labor to add to your quote. I highly recommend a detailed Scope of Work since Asterisk can do Anything and that is often used as part of the sales cycle. You need to outline exactly what features and functionality are included in the Scope of Work so you and your customer can tweak it so they do not expect the world. I find an hourly rate with a max is the safest way to price labor. The final Worksheet is a combination of Best Practices, testing and all of the items from your Scope of Work. The scope of work and the final worksheet checklist are to both be signed by you and your customer. I call this checklist the Customer Acceptance document which basically says you have delivered and tested what was expected, anything beyond that is Extra. Things like Rubber Feet Attached can obviously be omitted unless you use rubber feet. Anyways, that is the overview of what I have from several vendors. I am not sure where I have them archived but I will get the docs out there somewhere for download soon. Things like Rubber Feet Attached can obviously be omitted unless you use rubber feet. Thanks, Steve Totaro On Mon, May 5, 2008 at 7:43 PM, Roderick A. Anderson [EMAIL PROTECTED] wrote: Darren Wiebe wrote: If you're willing to cc me a copy I'll be in your debt. You bet. Rod -- Thanks, Darren Wiebe [EMAIL PROTECTED] Steve Totaro wrote: On Mon, May 5, 2008 at 5:10 PM, Roderick A. Anderson [EMAIL PROTECTED] wrote: Steve Totaro wrote: On Sun, May 4, 2008 at 1:55 PM, Roderick A. Anderson [EMAIL PROTECTED] wrote: Has anyone created a worksheet they can share for designing a dialplan, extensions, voicemail, etc. I'm making my way through the O'Reilly Book (dead tree version) and finding it enlightening. I have hacked at dialplans created by others but never actually came up with a design for my own system. It's sort of a work in progress made of bits and pieces from all over. Having a real plan would probably make things easier. Rod -- Rod, You will be glad that you are taking the learning curve plunge down the road. No pain, no gain. I can certainly say that I am glad I got into Asterisk way before there was any real documentation or GUIs for that matter. It forced me to learn the real deal Asterisk through trial and error which is invaluable if you plan on really getting into it. Then again, if you want easy, use a GUI. Easy isn't what I'm after. I was hoping for planning worksheets. Something to go over with a customer (I know I said this was for my personal system but that is the first step). How many extensions/ phones/ softphones, and what their /numeric/ extension will be. An IVR plan and the text that goes with it, voice-mail handling and mailboxes, etc. This type of stuff. So from the minimal number of responses -- yours :-) -- I'm going to guesstimate no one has anything like this at all or that they can or are able/willing to share. Out comes the notepad and the thinking cap. /-| Cheers, Rod -- Thanks, Steve Totaro Hey Rod, I think I may be able to help with worksheets from 3com, NEC, and other system vendor's sales channel. It obviously will not match exactly to Asterisk but will give you a great foundation for the functions and features that you need to question. I have my own but I prefer not to put it in the public domain. It is adapted from a conglomeration of many different proprietary systems that I have dealt with. I think many others have the same and consider it proprietary internal information for their business. Let me see what I can dig up from my archives. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by
[asterisk-users] DUNDi call impossible in one direction
Hello everybody, I've set up DUNDi between two asterisk boxes, and sometimes happens that calls from machine A can't reach peers in machine B, but calls from B to A work correctly. The strange thing is that the CLI command 'dundi show peers' shows correctly the registered peer in both servers, and in this situation if I make a call from B to A, suddenly peers in server A are able to call peers in machine B. Can anyone give me directions? Thanks in advance, -- Dr. Andrea Spadaccini Multimedia Technologies Institute - MTI S.r.l. Web: www.x-voice.it - Tel: +39 (0) 95 7224945 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis [EMAIL PROTECTED] wrote: lordfuknowsyou a écrit : Vinícius Fontes wrote: I use 1.4.18 with no problems. We have quite a few users(125 total between branches), but the call volume at the most has been around 15 active calls at a time. Any IAX2 phone or mostly SIP ? Do you use Call Queues ? We have less user than that, less concurrent call but quite a few crash/deadlocks Try SIP only if you can and report back. I think you will confirm what is pretty much a silent consensus (even among Digium Devs). Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX issues with 1.4.19.1
On Tue, 2008-05-06 at 08:42 -0500, Tilghman Lesher wrote: It's not actually a fix to the security fix. No, indeed. The security fix simply highlighted an issue which was already present in Asterisk. That may be true, but the security fix now depends on that new fix, so it's tangentially related at least. You would have seen this same issue prior to the security fix simply by running 'iax2 show channels', as that command runs through every single slot in the channel array, looking for active channels. The security issue fix simply made that slow search process (already there, nothing new) painfully obvious. Right. Which to me at least, tightly couples it. IOW, the security fix, while yes, it fixes the security problem, is quite useless without this other fix as it makes iax2 unstable. b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Basic modules of Asterisk
I just want to Run Asterisk with the basic required modules, What can I do to achieve so? My only requirement is to run SIP clients and the Dictate Module. Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
On Tue, May 06, 2008 at 01:38:37PM +0200, Benoit Plessis wrote: I'm wondering what version of asterisk people use in production environnement ? on which distribution ? And what is your setup like ? Well, we're running a cluster of about 15 boxes or so with Slack 10 or 12 and 1.2.17(?, either 14 or 17) and VICIdial. Yeah, you'd call it production. :-) It runs, knock on Formica-laminated particle board, pretty well. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
On Tue, May 6, 2008 at 9:35 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 06 May 2008 06:58:39 Steve Totaro wrote: While these may not be popular opinions, I still ask, what does SwitchVox use? What do some of the guys around here that setup large systems use? Is ABE even using 1.4 yet? Yes, ABE version C (in release for several months) is using the 1.4 codebase. All I see in the ABE release notes is 1.2 although I have heard that ABE should be running 1.4 Very Soon many many moons ago http://www.digium.com/en/docs/ABE/README . So either Digium doesn't trust 1.4 enough to use it for ABE or the README is out of date. The first clue should be that the copyright listed in that file is from 2006. Yes, it's very much out of date. -- Tilghman Does In Release equate to In the Wild or In Many Production Installations ? Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
Tilghman Lesher wrote: On Tuesday 06 May 2008 06:58:39 Steve Totaro wrote: All I see in the ABE release notes is 1.2 although I have heard that ABE should be running 1.4 Very Soon many many moons ago http://www.digium.com/en/docs/ABE/README . So either Digium doesn't trust 1.4 enough to use it for ABE or the README is out of date. The first clue should be that the copyright listed in that file is from 2006. Yes, it's very much out of date. Fix it! Beat some of those tech writers into submission! N. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
On Tue, May 06, 2008 at 10:01:54AM -0400, Jay R. Ashworth wrote: On Tue, May 06, 2008 at 01:38:37PM +0200, Benoit Plessis wrote: I'm wondering what version of asterisk people use in production environnement ? on which distribution ? And what is your setup like ? Well, we're running a cluster of about 15 boxes or so with Slack 10 or 12 and 1.2.17(?, either 14 or 17) and VICIdial. Yeah, you'd call it production. :-) Sorry: we're running zap channels at all the edges (Digium and Sangoma quad-T cards, primarily), and IAX2 in the middle. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
Benoit Plessis wrote: lordfuknowsyou a écrit : Vinícius Fontes wrote: I use 1.4.18 with no problems. We have quite a few users(125 total between branches), but the call volume at the most has been around 15 active calls at a time. Any IAX2 phone or mostly SIP ? Do you use Call Queues ? We have less user than that, less concurrent call but quite a few crash/deadlocks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users We use SIP and IAX2, we also do fax 2 email using spandsp and rx/txfax. I did have a problem with libpri during the upgrade and had to roll back to the one I was using prior. hth ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
Steve Totaro wrote: I use 1.4.18 with no problems. We have quite a few users(125 total between branches), but the call volume at the most has been around 15 active calls at a time. I would classify that as Light to Medium Call Volume or SMB. Let me clarify what I consider High Call Volume. ~400 simultaneous calls, all SIP or 95 on a box doing quad PRI to SIP gateway duty. 15k+ calls a day lasting an average of fifteen minutes. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I agree that it is SMB, never said I was a telco ;] just in production on 1.4.18. we do use sip,iax2 and pri. Our calls do last extended periods of time, especially when there are conferences. No call ques, and we do realtime voicemail,sip and iax to allow tennants web interfaces into the system through the standard 3 tiers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
Steve Totaro a écrit : On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis [EMAIL PROTECTED] wrote: Any IAX2 phone or mostly SIP ? Do you use Call Queues ? We have less user than that, less concurrent call but quite a few crash/deadlock Try SIP only if you can and report back. I think you will confirm what is pretty much a silent consensus (even among Digium Devs). Hi, that's what i was planning seeing all thoses answers. We initialy choosed IAX2 for the sendurl() support but i'll set-up a test periode in SIP-only to compare. Thanks, Steve Totaro Thanks to you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Performance issues
Hello, We are thinking in use asterisk-java to an billing solution, wich is the better choice, and if someone could give us a understandable description about the difference between DeadAGI and FastAGI, i found a very interesting project called asterisk2billing and they use DeadAGI, anyway wich one scale better? And there is a tool for test performance? Thanks, Roberto ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
We are using Asterisk 1.4.13 in production, were we have almost 30 SIP users on a Asterisk box, we are also using IAX to communicate between main Asterisk server and the other. we use Queues, Conference too. Regards, Sanjay Rajdev - Original Message - From: Benoit Plessis [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, May 6, 2008 5:08:37 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: [asterisk-users] Asterisk in Production ? Hi, I'm wondering what version of asterisk people use in production environnement ? on which distribution ? And what is your setup like ? We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 and it's quite unstable. We have ~30 IAX2 SoftPhones and encounter some Avoiding IAX destroy deadlock and now that we have added a Queue, it's worse than ever. The queue goes stuck quite often (agent are stuck in 'In use' state and if they logoff they can't log-in till an asterisk restart). regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
Tilghman Lesher a écrit : On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote: lordfuknowsyou a écrit : Vinícius Fontes wrote: I use 1.4.18 with no problems. We have quite a few users(125 total between branches), but the call volume at the most has been around 15 active calls at a time. Any IAX2 phone or mostly SIP ? Do you use Call Queues ? We have less user than that, less concurrent call but quite a few crash/deadlocks Have you reported these issues on the bugtracker? Well, the problem is finding usefull data to report. I've 4 core dumps thats show differents things: two seems to be related to ControlPlayback: #0 0xb705b658 in strcasecmp () from /lib/tls/libc.so.6 #1 0x0809c579 in ast_readframe () #2 0x0809defc in ast_streamfile () #3 0x0805e786 in ast_control_streamfile () #4 0xb698be5c in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #5 0x08298700 in ?? () #6 0xb470aec0 in ?? () #7 0xb698c1fc in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #8 0xb698c1fa in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #9 0x in ?? () One is pretty generic: #0 0x0809c9bc in ast_closestream () #1 0x08085d91 in ast_hangup () #2 0x080cd3d8 in pbx_builtin_setvar_helper () #3 0x080cf08e in ast_pbx_outgoing_exten () #4 0x080fde65 in ast_inet_ntoa () #5 0xb7eec560 in start_thread () from /lib/tls/libpthread.so.0 #6 0xb703667e in clone () from /lib/tls/libc.so.6 and the latest is thread/iax2 related: #0 0xb7ee71c7 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0 #1 0xb562a969 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so #2 0x0079 in ?? () #3 0x in ?? () #4 0xb547a148 in ?? () #5 0x080f0508 in ast_sched_add_variable () #6 0xb5647c89 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so #7 0x0012 in ?? () But my main problem is when the system just froze, it start mostly by the Queue not working anymore, with member stuck in 'in use' stack (should not happen with IAX2 agent IIRC, given that we had to build macros using GROUP() to detect in use IAX2 agent) Then the console (asterisk -rcTvvv) start to freeze (completion doesn't work, message stop from being displayed and even command output is lost). And i'm reading http://www.asterisk.org/developers/bug-guidelines which speak of using SVN trunk version of asterisk, thing i'm not really eager to try on a live system... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
Hello, Our company did 200+ installations around the globe and had no issues with stability with correct Asterisk version. We used most of 1.4. As far as I remember 1.4.16 has some nasty bugs along with 1.4.19.x (SIP + realtime). So current stable is 1.4.18.1 (for us). For load check: http://wiki.kolmisoft.com/index.php/How_fast_MOR_can_perform It shows how our billing application performs on top of Asterisk (2049 channels) and we can push it even further with some improvements. We DO NOT RESTART our Asterisk installations daily or weekly. They work for months. Regards, Mindaugas Kezys http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Benoit Plessis Sent: Tuesday, May 06, 2008 2:39 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk in Production ? Hi, I'm wondering what version of asterisk people use in production environnement ? on which distribution ? And what is your setup like ? We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 and it's quite unstable. We have ~30 IAX2 SoftPhones and encounter some Avoiding IAX destroy deadlock and now that we have added a Queue, it's worse than ever. The queue goes stuck quite often (agent are stuck in 'In use' state and if they logoff they can't log-in till an asterisk restart). regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
I'm using 1.4.18 in production on 2 boxes... one of which being a custom built desktop basically, the other being a Dell 1950 III We are in a migration phase to the Dell box, right now the 1st box is doing nothing more than being a PSTN gateway to some FXO lines... basically waiting for numbers to be ported off the analog lines and onto the new T1 which is connected to the Dell box. We have the 2 boxes connected by IAX2 trunk. I had 1.4.19 and 1.4.19.1 running on the Dell box, but it started giving me a lot of trouble with the IAX2 trunk, the trunk would (seemingly) go into UNREACHABLE status and never come back without restarting asterisk (reload, or iax2 reload wouldn’t cut it). Also, occasionally people trying to make outbound calls (and this probably happened on inbound as well), would get a all circuits are busy message because of the IAX2 channel driver reporting congestion on the trunk even though it was up (and not congested) Unfortunately as this is a production box I didn’t really have time to try and debug it so I simply downgraded to .18 since it has proven itself well on the 1st box. So far since I;ve downgraded to .18 I haven’t had any problems. Both installs I have running ontop of Gentoo (wouldn’t recommend it if you are new to Linux or don’t like tweak-ability). That all being said, I'll probably give .20 a try when its released, as I see there have been some IAX2 bug fixes in it... but also by the time .20 is released I probably will have retired the box being used as a PSTN gateway and won’t need the IAX2 trunk anymore. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vinícius Fontes Sent: Tuesday, May 06, 2008 8:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk in Production ? There were some really unstable Asterisk releases in the 1.4 branch. I personally use 1.4.13 or 1.4.15 in production. Every single time I tried 1.4.16 or higher I had problems. Att Vinícius Fontes Desenvolvimento Canall Tecnologia em Comunicações Ltda. - Steve Totaro [EMAIL PROTECTED] escreveu: On Tue, May 6, 2008 at 7:38 AM, Benoit Plessis [EMAIL PROTECTED] wrote: Hi, I'm wondering what version of asterisk people use in production environnement ? on which distribution ? And what is your setup like ? We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 and it's quite unstable. We have ~30 IAX2 SoftPhones and encounter some Avoiding IAX destroy deadlock and now that we have added a Queue, it's worse than ever. The queue goes stuck quite often (agent are stuck in 'In use' state and if they logoff they can't log-in till an asterisk restart). regards I am personally a proponent of Asterisk 1.2.X as I see more and more fatal bugs in the 1.4.X code come up on the lists as well as IAX2 bugs. I constantly hear Asterisk 1.4.whatever is much better, but the bugs coming out are not just unexpected behavior that one could live with, they are segfaults, system crashes, modules not getting installed (Zaptel). I use SIP since I have seen quite a few issues with IAX2 that were solved by simply switching to SIP. The above two yield solid systems under heavy load for me. OS is not so important I do not believe. I have some running FC8 and more running CentOS, both rock solid. I think the general consensus on OS is use what you are most familiar with. While these may not be popular opinions, I still ask, what does SwitchVox use? What do some of the guys around here that setup large systems use? Is ABE even using 1.4 yet? All I see in the ABE release notes is 1.2 although I have heard that ABE should be running 1.4 Very Soon many many moons ago http://www.digium.com/en/docs/ABE/README . So either Digium doesn't trust 1.4 enough to use it for ABE or the README is out of date. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Performance issues
Google is awesome http://www.voip-info.org/wiki-Asterisk+AGI -- Matt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of chetherston miles Sent: Tuesday, May 06, 2008 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Performance issues Hello, We are thinking in use asterisk-java to an billing solution, wich is the better choice, and if someone could give us a understandable description about the difference between DeadAGI and FastAGI, i found a very interesting project called asterisk2billing and they use DeadAGI, anyway wich one scale better? And there is a tool for test performance? Thanks, Roberto ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call manager using Asterisk as voicemail server (SIP) not working ...
These are the instructions that I followed. I did managed to get the
fast busy to go away, but the RDNIS simply does not seem to work. These
are the instructions that I followed on this project. I have run out of
time trying to get Call Manager 4.x to talk to Asterisk 1.4.
http://www.voip-info.org/wiki/index.php?page_id=2596#editcomments
These instructions although a good start, simply lack the pictures or
images to set up CCM properly, and because of the coding change from
earlier versions, this just doesn't seem to allow voice mail to work.
I have learned a lot about asterisk, but am frustrated by this
experience.
Thanks Sean for the info about the change of the rdnis command format.
Kind regards,
Steve
On Mon, 2008-05-05 at 23:33 -0400, Steve Hickel wrote:
Sean,
Here is what I changed. Now I have a fast busy...
Steve
[demo]
exten=s,1,Wait(1)
exten=s,n,Answer
exten=s,n,Set(TIMEOUT(digit)=5)
exten=s,n,Set(TIMEOUT(response)=10)
exten=s,n(restart),BackGround(demo-congrats)
exten=s,n(instruct),BackGround(demo-instruct)
exten=s,n,WaitExten
exten=2,1,BackGround(demo-moreinfo)
exten=2,n,Goto(s,instruct)
exten=3,1,Set(LANGUAGE()=fr)
exten=3,n,Goto(s,restart)
exten=1000,1,Goto(default,s,1)
exten=1234,1,Playback(transfer,skip)
exten=1234,n,Macro(stdexten,1234,${GLOBAL(CONSOLE)})
exten=1235,1,Voicemail(1234,u)
exten=1236,1,Dial(Console/dsp)
exten=1236,n,Voicemail(1234,b)
exten=#,1,Playback(demo-thanks)
exten=#,n,Hangup
exten=t,1,Goto(#,1)
exten=i,1,Playback(invalid)
exten=500,1,Playback(demo-abouttotry)
exten=500,n,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
exten=500,n,Playback(demo-nogo)
exten=500,n,Goto(s,6)
exten=600,1,Playback(demo-echotest)
exten=600,n,Echo
exten=600,n,Playback(demo-echodone)
exten=600,n,Goto(s,6)
exten=76245,1,Macro(page,SIP/Grandstream1)
exten=_7XXX,1,Macro(page,SIP/${EXTEN})
exten=7999,1,Set(TIMEOUT(absolute)=60)
exten=7999,2,Page(Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED]Local/[EMAIL PROTECTED]/n|d)
exten=,1,VoicemailMain
exten=,n,Goto(s,6)
[general]
static=yes
writeprotect=no
clearglobalvars=no
autofallthrough=yes
priorityjumping=no
[default]
exten=_230,1,SetCallerID(${EXTEN:3})
exten=_230,2,Dial(SIP/[EMAIL PROTECTED])
exten=_230,3,Answer
exten=_230,4,Wait,1
exten=_230,5,Hangup
exten=_231,1,SetCallerID(${EXTEN:3})
exten=_231,2,Dial(SIP/[EMAIL PROTECTED])
exten=_231,3,Answer
exten=_231,4,Wait,1
exten=_231,5,Hangup
exten=,1,VoiceMailMain
[incoming]
exten=,1,GotoIf($[${CALLERID(rdnis)}]?2:400)
exten=,2,MailboxExists(${CALLERID(rdnis)[EMAIL PROTECTED])
exten=,3,Congestion
exten=,103,Voicemail(su${CALLERID(rdnis)}
exten=,104,Playback(vm-goodbye)
exten=,105,Hangup
exten=,400,VoicemailMain
__
From: Sean Dennis [mailto:[EMAIL PROTECTED]
To: [EMAIL PROTECTED], Asterisk Users Mailing List -
Non-Commercial Discussion
[mailto:[EMAIL PROTECTED]
Sent: Mon, 05 May 2008 17:58:32 -0400
Subject: Re: [asterisk-users] Call manager using Asterisk as
voicemail server (SIP) not working ...
Steve Hickel wrote:
I have sip set up on Callmanager 4.x. When others call my
ext of 2016 on
ccm after a busy or no answer, asterisk voice mail answers
by saying,
Mailbox password. I want it to put them into my
mailbox so they
can leave a message. Somehow I must be missing something...
Please
help!
I have spent 19 hours easy on trying to figure this one
out.
SIP DN is on CCM
VOICEMAIL on Asterisk is .
Here is my sip.conf:
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
allowexternaldomains=yes
allowexternalinvites=no
allowguest=yes
allowsubscribe=no
allowtransfer=yes
alwaysauthreject=no
autodomain=no
callevents=no
compactheaders=no
dumphistory=no
g726nonstandard=no
ignoreregexpire=no
jbenable=no
jbforce=no
jblog=no
maxcallbitrate=384
maxexpiry=3600
minexpiry=60
nat=no
notifyringing=no
pedantic=no
promiscredir=no
recordhistory=no
relaxdtmf=no
rtcachefriends=no
rtsavesysname=no
rtupdate=no
sendrpid=yes
Re: [asterisk-users] Asterisk in Production ?
On Tuesday 06 May 2008 09:30:50 Benoit Plessis wrote: Tilghman Lesher a écrit : On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote: lordfuknowsyou a écrit : Vinícius Fontes wrote: I use 1.4.18 with no problems. We have quite a few users(125 total between branches), but the call volume at the most has been around 15 active calls at a time. Any IAX2 phone or mostly SIP ? Do you use Call Queues ? We have less user than that, less concurrent call but quite a few crash/deadlocks Have you reported these issues on the bugtracker? Well, the problem is finding usefull data to report. I've 4 core dumps thats show differents things: two seems to be related to ControlPlayback: #0 0xb705b658 in strcasecmp () from /lib/tls/libc.so.6 #1 0x0809c579 in ast_readframe () #2 0x0809defc in ast_streamfile () #3 0x0805e786 in ast_control_streamfile () #4 0xb698be5c in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #5 0x08298700 in ?? () #6 0xb470aec0 in ?? () #7 0xb698c1fc in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #8 0xb698c1fa in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #9 0x in ?? () I'd love to see a 'bt full' on this one. One is pretty generic: #0 0x0809c9bc in ast_closestream () #1 0x08085d91 in ast_hangup () #2 0x080cd3d8 in pbx_builtin_setvar_helper () #3 0x080cf08e in ast_pbx_outgoing_exten () #4 0x080fde65 in ast_inet_ntoa () #5 0xb7eec560 in start_thread () from /lib/tls/libpthread.so.0 #6 0xb703667e in clone () from /lib/tls/libc.so.6 Ditto, bt full. and the latest is thread/iax2 related: #0 0xb7ee71c7 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0 #1 0xb562a969 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so #2 0x0079 in ?? () #3 0x in ?? () #4 0xb547a148 in ?? () #5 0x080f0508 in ast_sched_add_variable () #6 0xb5647c89 in ?? () from /usr/lib/asterisk/modules/chan_iax2.so #7 0x0012 in ?? () This one may need valgrind to track down the problem, but please be sure to run 1.4.18 or later, as we've already fixed a problem that produced backtraces similar to this. But my main problem is when the system just froze, it start mostly by the Queue not working anymore, with member stuck in 'in use' stack (should not happen with IAX2 agent IIRC, given that we had to build macros using GROUP() to detect in use IAX2 agent) Then the console (asterisk -rcTvvv) start to freeze (completion doesn't work, message stop from being displayed and even command output is lost). And i'm reading http://www.asterisk.org/developers/bug-guidelines which speak of using SVN trunk version of asterisk, thing i'm not really eager to try on a live system... I don't see anywhere on that page that recommends that you try SVN trunk, only the latest SVN (which is probably confusing, but what is meant is to try the latest SVN in the 1.4 branch, which is the release branch. Release candidates and releases are tagged directly off that branch). -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
On Tuesday 06 May 2008 09:02:47 Steve Totaro wrote: On Tue, May 6, 2008 at 9:35 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 06 May 2008 06:58:39 Steve Totaro wrote: While these may not be popular opinions, I still ask, what does SwitchVox use? What do some of the guys around here that setup large systems use? Is ABE even using 1.4 yet? Yes, ABE version C (in release for several months) is using the 1.4 codebase. Does In Release equate to In the Wild or In Many Production Installations ? I sense that there are quite a few people who are running version C and a few holdouts still running B, but that's based on a wet-finger-in-the-wind estimation, not on any industry surveys. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production
On May 6, 2008, at 10:20 AM, [EMAIL PROTECTED] wrote: I'm wondering what version of asterisk people use in production environnement ? on which distribution ? And what is your setup like ? We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 and it's quite unstable. I'm running 1.4.19 and it has been pretty stable. Anything before 1.4.19, however, I found was embarrassingly unstable. I'd often get several crashes within an hour. However, since moving to 19 things have been better. I don't run Queues, though, but I do run a custom derivative of Queues that fixed some bugs and greatly enhanced its usability for us. We do tens of thousands of calls per day (mostly inbound) running on under Debian, although I had to upgrade the kernel to 2.6.23.11 in order to get ztdummy to work on my HP DL380. CPU load remains rather low. We are all SIP, no zaptel. I used to run IAX2 between my three servers (one's a backup and for testing, the other handles desk phones and ATAs), but found IAX2 very, very unreliable. It would hang Asterisk, crash, etc. I just replaced it with SIP (and turned off the module) and those problems went away. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
Tilghman Lesher a écrit : On Tuesday 06 May 2008 09:30:50 Benoit Plessis wrote: Tilghman Lesher a écrit : On Tuesday 06 May 2008 08:18:07 Benoit Plessis wrote: lordfuknowsyou a écrit : Vinícius Fontes wrote: I use 1.4.18 with no problems. We have quite a few users(125 total between branches), but the call volume at the most has been around 15 active calls at a time. Any IAX2 phone or mostly SIP ? Do you use Call Queues ? We have less user than that, less concurrent call but quite a few crash/deadlocks Have you reported these issues on the bugtracker? Well, the problem is finding usefull data to report. I've 4 core dumps thats show differents things: two seems to be related to ControlPlayback: #0 0xb705b658 in strcasecmp () from /lib/tls/libc.so.6 #1 0x0809c579 in ast_readframe () #2 0x0809defc in ast_streamfile () #3 0x0805e786 in ast_control_streamfile () #4 0xb698be5c in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #5 0x08298700 in ?? () #6 0xb470aec0 in ?? () #7 0xb698c1fc in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #8 0xb698c1fa in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so #9 0x in ?? () I'd love to see a 'bt full' on this one. Here it is, but since the AsteriskNow release has stripped the binary i fear it won't be of much use: #0 0xb705b658 in strcasecmp () from /lib/tls/libc.so.6 No symbol table info available. #1 0x0809c579 in ast_readframe () No symbol table info available. #2 0x0809defc in ast_streamfile () No symbol table info available. #3 0x0805e786 in ast_control_streamfile () No symbol table info available. #4 0xb698be5c in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #5 0x08298700 in ?? () No symbol table info available. #6 0xb470aec0 in ?? () No symbol table info available. #7 0xb698c1fc in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #8 0xb698c1fa in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #9 0x in ?? () No symbol table info available. #10 0x in ?? () No symbol table info available. #11 0x in ?? () No symbol table info available. #12 0x0bb8 in ?? () No symbol table info available. #13 0x2f727669 in ?? () No symbol table info available. #14 0x65696c63 in ?? () No symbol table info available. #15 0x2f73746e in ?? () No symbol table info available. #16 0x6a6e6f62 in ?? () No symbol table info available. #17 0x2d72756f in ?? () No symbol table info available. #18 0x6e656962 in ?? () No symbol table info available. #19 0x756e6576 in ?? () No symbol table info available. #20 0x6568632d in ?? () No symbol table info available. #21 0x6f702d7a in ?? () No symbol table info available. #22 0x62726577 in ?? () No symbol table info available. #23 0x6974756f in ?? () No symbol table info available. #24 0x2d657571 in ?? () No symbol table info available. #25 0x76726573 in ?? () No symbol table info available. #26 0x73656369 in ?? () No symbol table info available. #27 0x696c632d in ?? () No symbol table info available. #28 0x00746e65 in ?? () No symbol table info available. #29 0x0001 in ?? () No symbol table info available. #30 0xb470af20 in ?? () No symbol table info available. #31 0x081aa084 in ?? () No symbol table info available. #32 0x001b in ?? () No symbol table info available. #33 0x0025 in ?? () No symbol table info available. #34 0x0028 in ?? () No symbol table info available. #35 0xb698c1fc in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #36 0x in ?? () No symbol table info available. #37 0xb698c1fa in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #38 0x0829c4a8 in ?? () No symbol table info available. #39 0x0bb8 in ?? () No symbol table info available. #40 0x in ?? () No symbol table info available. #41 0xb470aec0 in ?? () No symbol table info available. #42 0x in ?? () No symbol table info available. #43 0xb698c1fc in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #44 0xb698c1fa in ?? () from /usr/lib/asterisk/modules/app_controlplayback.so No symbol table info available. #45 0x in ?? () No symbol table info available. #46 0x in ?? () No symbol table info available. #47 0x in ?? () No symbol table info available. #48 0x in ?? () No symbol table info available. #49 0x08298700 in ?? () No symbol table info available. #50 0xb705b631 in strcasecmp () from /lib/tls/libc.so.6 No symbol table info available. #51 0x080c8740 in pbx_substitute_variables_helper () No symbol table info available. #52 0x080cd170 in pbx_builtin_setvar_helper () No symbol table info available. #53 0x080cf08e in ast_pbx_outgoing_exten () No symbol
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
Tilghman Lesher wrote: On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote: 5 maj 2008 kl. 19.58 skrev Tilghman Lesher: On Monday 05 May 2008 11:24, Johansson Olle E wrote: 5 maj 2008 kl. 17.51 skrev Tilghman Lesher: On Monday 05 May 2008 09:45, Johansson Olle E wrote: Another issue that we need to fix with the MYSQL driver is that we're lacking a connection pool. Everything seems to be handled over one connection to Mysql, which causes issues. That's not true. The MYSQL app generally uses multiple connections, one for each channel. The only way one might use only a single connection is by using a global variable to store a single connection id, but that method is not documented anywhere, AFAIK. You talk about the Mysql APP, but is this the case with the Realtime driver as well? No, the native Realtime driver uses a single connection. The ODBC Realtime driver generally uses a single connection but can be configured to use a separate connection for each query. So, we're back to where we started. A developer that can help us with a connection pool or a separate connection for each query would be a Nice Thing (TM). What issues are you specifically seeing that merit using multiple connections? I can specify an issue that would merit multiple connections, if the link to your db goes away Asterisk likes to freeze writing CDRs. I have a few remote * servers that this happens to. My solution so far has been to record CDR's to a local DB and then have a perl script that attempts to move them over to my transaction DB. I would suggest this solution to anyone who depends on their CDR records. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
Steve Totaro a écrit : On Tue, May 6, 2008 at 9:18 AM, Benoit Plessis [EMAIL PROTECTED] wrote: lordfuknowsyou a écrit : Vinícius Fontes wrote: I use 1.4.18 with no problems. We have quite a few users(125 total between branches), but the call volume at the most has been around 15 active calls at a time. Any IAX2 phone or mostly SIP ? Do you use Call Queues ? We have less user than that, less concurrent call but quite a few crash/deadlocks Try SIP only if you can and report back. I think you will confirm what is pretty much a silent consensus (even among Digium Devs). Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I've tried SIP only but i already got one 'stuck' Queue member: Members: Local/[EMAIL PROTECTED] with penalty 10 (dynamic) (In use) has taken 1 calls (last was 45 secs ago) Local/[EMAIL PROTECTED] with penalty 20 (dynamic) (Not in use) has taken no calls yet Callers: 1. Zap/10-1 (wait: 0:18, prio: 0) [May 6 17:48:35] NOTICE[2047]: app_queue.c:2152 wait_for_answer: No one is answering queue 'support' (1/0/0) asterix*CLI core show channels Channel Location State Application(Data) SIP/rtournier-081ef2 (None) Up Bridged Call(Local/[EMAIL PROTECTED] but the other end of the bridged call is long gone ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI D-Channel reconfiguration = crash asterisk?
Hello, I just had to have MTS Allstream fix a new T1 install that we have that we aren't running in production yet, but it is attached to a production machine. Apparently they setup the T1 with only a 1 B-channel (how useful!) even though we had ordered it fully loaded with 23. Anyways... they just reconfigured the T1 to activate all the T1 channels and this is what I got on my * console: == Primary D-Channel on span 1 down nelson*CLI Disconnected from Asterisk server ^^ asterisk crashed. Unfortunately I didn't have * setup on this box to dump a core file, so the only additional debug info I can provide is from my asterisk log file: [May 6 11:42:23] VERBOSE[16656] logger.c: == Primary D-Channel on span 1 down [May 6 11:42:23] WARNING[16656] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [May 6 11:42:23] WARNING[16656] chan_zap.c: The PRI Call have not been destroyed Those are they only 3 relevant lines in the log file. -- Matt Disclaimer Statement: This e-mail is confidential and is intended for the above-named recipient(s) only. If you are not the intended recipient and/or have received this e-mail in error, please notify us by telephone and delete this e-mail from your system without retaining a copy in any form. Any unauthorized use or disclosure of this e-mail is prohibited. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI D-Channel reconfiguration = crash asterisk?
My bad, I also should of mentioned... That was on Asterisk 1.4.18 and Zaptel 1.4.10 Using a TE220B -- Matt From: Matt Watson Sent: Tuesday, May 06, 2008 12:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: PRI D-Channel reconfiguration = crash asterisk? Hello, I just had to have MTS Allstream fix a new T1 install that we have that we aren't running in production yet, but it is attached to a production machine. Apparently they setup the T1 with only a 1 B-channel (how useful!) even though we had ordered it fully loaded with 23. Anyways... they just reconfigured the T1 to activate all the T1 channels and this is what I got on my * console: == Primary D-Channel on span 1 down nelson*CLI Disconnected from Asterisk server ^^ asterisk crashed. Unfortunately I didn't have * setup on this box to dump a core file, so the only additional debug info I can provide is from my asterisk log file: [May 6 11:42:23] VERBOSE[16656] logger.c: == Primary D-Channel on span 1 down [May 6 11:42:23] WARNING[16656] chan_zap.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [May 6 11:42:23] WARNING[16656] chan_zap.c: The PRI Call have not been destroyed Those are they only 3 relevant lines in the log file. -- Matt Disclaimer Statement: This e-mail is confidential and is intended for the above-named recipient(s) only. If you are not the intended recipient and/or have received this e-mail in error, please notify us by telephone and delete this e-mail from your system without retaining a copy in any form. Any unauthorized use or disclosure of this e-mail is prohibited. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall - How to automatically block collect calls
I`m using collect call blocking with astunicall from moises and it`s working properly. UC_CATEGORY=INTERNATIONAL_DATA (for brazilian users) indicates a collect call. I spent a long time searching a way to do it, but it was only possible with moises code. Thank you. Luis A P Barbosa. 2008/5/6 Oscar Patricio [EMAIL PROTECTED]: Hi! Thank you for your answer. However I would like to know, if there is any other possibility of making this work, with older unicall versions. The reason I ask, is because it's not easy for me to upgrade the asterisk and unicall version on my machine, since it is a production machine, and I would consider a software upgrade really only if no other solution is possible. Thanks in advance for your answer. Best regards, Óscar Patrício Moises Silva escreveu: The latest version of the driver included in http://www.moythreads.com/astunicall/ comes with a change that will set the variable UC_CATEGORY in your dialplan, Brasil has a special category for those calls, don't remember the name that will show up, but you can make a couple of tests and then drop any call with that specific category. Moy On Mon, May 5, 2008 at 9:14 AM, Oscar Patricio [EMAIL PROTECTED] wrote: Hi! I am using asterisk with unicall in brasil. Everything was working fine, but now we want to set up a way to automatically drop collect calls, because we have an IVR answering all calls automatically! Can you tell me, what I have to configure to block collect calls in the asterisk? Thank you! Best regards, Óscar Patrício ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] finding Asterisk user group users and enthusiasts in the (Salt Lake City Utah, Chicago IL, Boston MA, Tampa FL)
Hi all, Are there Asterisk user groups and organized Asterisk enthusiasts in the following cities: Salt Lake City Utah, Chicago IL, Boston MA, Tampa FL? Looking to meet up with folks in these cities during the month of May when I am in those cities. Drop me a mail if you are looking to shoot the breeze and discuss all topics asterisk unified communications. Thanks Ming -- Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] This e-mail is confidential ... (was: Re: PRI D-Channel reconfiguration = crash asterisk?)
Matt Watson schrieb: Disclaimer Statement: This e-mail is confidential and is intended for the above-named recipient(s) only. If you are not the intended recipient and/or have received this e-mail in error, please notify us by telephone and delete this e-mail from your system without retaining a copy in any form. Any unauthorized use or disclosure of this e-mail is prohibited. Your confidential e-mail is going to end up on Google ... Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Performance issues
I'm not an expert, but FastAGI you use in a live channel over TCP connection ... Most people agree it is better because doesn't spawn a new process every time it is called ... but they also suggest to do not use in the same server ... DeadAGI you cannot use it in a live channel ... By the way, a2billing is a nice software ... Areski does a great job ... I neither use it a lot nor in a production high volume environment but I liked it ... - Original Message - Subject: [asterisk-users] Performance issues From: chetherston miles ;[EMAIL PROTECTED] Date: Tue, May 6, 2008 10:20 Hello, We are thinking in use asterisk-java to an billing solution, wich is the better choice, and if someone could give us a understandable description about the difference between DeadAGI and FastAGI, i found a very interesting project called asterisk2billing and they use DeadAGI, anyway wich one scale better? And there is a tool for test performance? Thanks, Roberto ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mixmonitor recording issue
Hi All I am using asterisk 1.4.18.1.My problem is that Mixmonitor is not recording complete recording. Suppose i got a call connected and talking for 3 minutes then mixmonitor records only 2 minutes of call. This problem happens randomly.Please help as i am suffering very much due to this problem. Rahul Yadav ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Asterisk as root
Really not.. if only you delete sh, yes, but i say make a symlink from /bin/bash to /bin/sh Ubuntu 7.04 and above, comes with the shell dash as sh, ubuntu 6.06 comes with bash as sh, I got problems to start daemon, when sh points to dash.. safe_asterisk don's start... I read 1.4.19 don't need anymore safe_asterisk, but, what about if I need 1.2.x ? or 1.4.18 ? I talk for example if I use unicall for E1 (MFCR2) when I need that versions... someone do? Oh!!! Now understand... I forget it... when I say about delete sh... i forget say that is necessary, to create a symlink from /bin/bash to /bin/sh I'm so sorry :$ -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Tzafrir Cohen Enviado el: Lunes, 05 de Mayo de 2008 07:35 p.m. Para: asterisk-users@lists.digium.com Asunto: Re: [asterisk-users] Running Asterisk as root On Mon, May 05, 2008 at 07:18:08PM -0500, Cesar Benjamin Garcia Martinez wrote: Btw... delete the symlink sh - dash into /bin BAD THAT BREAKS THE SYSTEM (leaves it without /bin/sh, making half the scripts fail) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 3078 (20080506) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Asterisk as root
Hum. About the /var/run i do thats changes in the conf and the creation fo /var/run into /var/lib/asterisk becouse Works :P. Yes, Ubuntu cleans al into /var/run and that's my solution, I believe is possible touch something in daemon for do work fine but I consider more simple make 2 folders and modify one line Maybe, the init.d script works well if comes from official package, I never has installed asterisk from package, I prefer from sources. On Mon, May 05, 2008 at 07:18:08PM -0500, Cesar Benjamin Garcia Martinez wrote: Move to root: sudo -s type your passwd and as root: Edit the file /etc/init.d/asterisk And uncommet the two lines than sasys something like AST_USER=asterisk AST_GROUP=asterisk You need to create the user asterisk on your system. And create another symlink sh to bash: cd /bin rm -f sh ln -s bash sh Why is that? Debian / Ubuntu policy is that a script that is not posix sh should use /bin/bash. Any script of Asterisk does not fit the policy and has not bit shot^Wfixed yet? The fix is to edit the ofending script: #!/bin/sh - #!/bin/bash Edit your /etc/asterisk/asterisk.conf and replace the line: astrundir = /var/run With: astrundir = /var/lib/asterisk/var/run /var/run/asterisk Everything under /var/run is deleted at boot with Ubuntu, so the init.d script should recreate that directory and give it proper permissions if it does not exist. (or use the one from the Asterisk package) Create that folder: mkdir -p /var/lib/asterisk/var/run /var/run/asterisk, as mentioned above. and it should be created in the init.d script . and, chown to asterisk:asterisk the folders: /var/lib/asterisk/ /usr/lib/asterisk/ No real need for /usr/lib/asterisk to be owned by Asterisk. It is read-only. /usr is read-only, as you recall. /var/log/asterisk/ chown -Rv asterisk:asterisk /var/lib/asterisk/ # chown -Rv asterisk:asterisk /usr/lib/asterisk/ chown -Rv asterisk:asterisk /var/log/asterisk/ that's all Btw... delete the symlink sh - dash into /bin NOT Start daemon /etc/init.d/asterisk start -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 3078 (20080506) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how do I set callerid for incoming iax?
Using 1.6-rc8.
In iax.conf on the calling box, I have:
[iax-out]
.
callerid = sean 447
I even also put the same on called box.
But I can't seem to set the callerid:
exten =_NXX,1,Answer()
exten =_NXX,n,NoOp(${CALLERID(num)})
Answer(IAX2/iax-in-7, ) in new stack
NoOp(IAX2/iax-in-7, ) in new stack
So how do I set callerid?
sean
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Re: [asterisk-users] Mixmonitor recording issue
I had a similar problem. In my case Asterisk was crashing due to MixMonitor() and then automatically restarting. I have never found a alternative solution to record the calls. Regards, Sanjay Rajdev - Original Message - From: Rahul Yadav [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, May 6, 2008 10:54:32 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: [asterisk-users] Mixmonitor recording issue Hi All I am using asterisk 1.4.18.1.My problem is that Mixmonitor is not recording complete recording. Suppose i got a call connected and talking for 3 minutes then mixmonitor records only 2 minutes of call. This problem happens randomly.Please help as i am suffering very much due to this problem. Rahul Yadav ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] This e-mail is confidential ... (was: Re: PRI D-Channel reconfiguration = crash asterisk?)
That's fine... honestly I hate the message myself, however corporate policy is corporate policy so there isn't much of a point in discussing it. That being said, the message does clearly say that the message is for the named recipients, in this particular case, the named recipient is a public mailing list. By my action of sending a message to a public mailing list, one can say there is implied consent that it gets distributed to whomever the mailing list chooses on my behalf. Thanks, -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen Sent: Tuesday, May 06, 2008 12:27 PM To: Asterisk Users Subject: [asterisk-users] This e-mail is confidential ... (was: Re: PRI D-Channel reconfiguration = crash asterisk?) Matt Watson schrieb: Disclaimer Statement: This e-mail is confidential and is intended for the above-named recipient(s) only. If you are not the intended recipient and/or have received this e-mail in error, please notify us by telephone and delete this e-mail from your system without retaining a copy in any form. Any unauthorized use or disclosure of this e-mail is prohibited. Your confidential e-mail is going to end up on Google ... Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What is field 'User/ANR'
*1.4 Sorry for a dumb question, but I'm working with my SIP provider on a problem and I can't answer this question for them. They don't know Asterisk. When I do a sip show channels What is the User/ANR field? Bill ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how do I set callerid for incoming iax?
On Tuesday 06 May 2008 12:45:42 sean darcy wrote:
Using 1.6-rc8.
In iax.conf on the calling box, I have:
[iax-out]
.
callerid = sean 447
I even also put the same on called box.
But I can't seem to set the callerid:
exten =_NXX,1,Answer()
exten =_NXX,n,NoOp(${CALLERID(num)})
Answer(IAX2/iax-in-7, ) in new stack
NoOp(IAX2/iax-in-7, ) in new stack
So how do I set callerid?
iax-out != iax-in
--
Tilghman
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Re: [asterisk-users] Mixmonitor recording issue
I had a similar problem when the calls were not recorded when there was a
transfer. Is that your case? If so, the solution is to start recording on the
inbound leg of the call for all channels. Just like that:
[default]
exten = _00[2-6]XXX,1,Dial(Local/[EMAIL PROTECTED])
[recording]
exten =
_00[2-6]XXX,1,MixMonitor(${CALLERID(num)}-${STRFTIME(${EPOCH},America/Sao_Paulo,%Y-%m-%d-%H-%M-%S)}-${EXTEN}.wav)
exten = _00[2-6]XXX,n,Dial(Zap/g1/${EXTEN:1})
Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações Ltda.
- Rahul Yadav [EMAIL PROTECTED] escreveu:
Hi All
I am using asterisk 1.4.18.1.My problem is that Mixmonitor is not
recording complete recording.
Suppose i got a call connected and talking for 3 minutes then
mixmonitor records only 2 minutes of call.
This problem happens randomly.Please help as i am suffering very much
due to this problem.
Rahul Yadav
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Re: [asterisk-users] Asterisk in Production ?
On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote: Here it is, but since the AsteriskNow release has stripped the binary i fear it won't be of much use: Is there any -debug package for asterisknow's asterisk package? On RedHat they are generated automatically. On Debian they require some extra settings, and has been present in recent Asterisk packages (the asterisk-dbg package) but not in all of the smaller modules packages. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk queue cluster
I setup two asterisk servers with identical settings (same extensions, same queues, etc). Each one is connected to the same amount of incoming/outgoing links (1 PRI, 4 BRI, 1 IAX friend, etc, on each box). Most extensions are sip and they register via DNS SRV and other methods so that the two servers are load balanced. Incoming PSTN calls (BRI) reach 50% each server so that's load balanced too. I use DUNDi with a IAX friend between the 2 servers to lookup where an extension was registered (whether on pbx1 or pbx2). Everything is working as expected (SIP on PBX1 calls SIP on PBX2 and vice versa; PSTN incoming call reaches SIP extension or group whether it registered on pbx1 or pbx2, etc.). My question regards queues. Incoming PSTN calls have 50% chance of entering, say, queue number 1000 on pbx1 and 50% on pbx2 (same queue 1000). All calls will be correctly bridged to the agent, whether it randomly registered on pbx1 or pbx2. However, how can I make pbx1 and pbx2 counters consistent? ie. first call enters queue 1000 on pbx1 and will have position 1; second call entes queue 1000 and happens to fall on pbx2 and will also have position 1. The worst case would be if there are, say, 10 callers in queue 1000 on pbx1 and the 11th call arrives on pbx2 with position 1. Is there a way of coherently setting up a clustered queue? Does anyone have examples/workarounds/links? Thanks! Vieri Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Storing voicemail on samba share
A client has asked that our asterisk installation leverage their large investment in their existing data center infrastructure. We're thinking about putting the voicemail messages onto a Samba share (on their file servers). Any pros/cons to this? Does network/samba latency create choppiness? Thanks, MD ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
Tzafrir Cohen a écrit : On Tue, May 06, 2008 at 05:37:09PM +0200, Benoit Plessis wrote: Here it is, but since the AsteriskNow release has stripped the binary i fear it won't be of much use: Is there any -debug package for asterisknow's asterisk package? On RedHat they are generated automatically. On Debian they require some extra settings, and has been present in recent Asterisk packages (the asterisk-dbg package) but not in all of the smaller modules packages. Nope, already tried this before posting but nothing like that appears on conary anyway, i'll be migrating on a debian asap, since i now this much better and the advantages of AsteriskNow keep reducing as a matter of fact i already now that some thing that doesn't work under AstNow (my siemens sip hardphones, and my SIP provider (Keyyo) at least) work with the debian packaged asterisk. Well for the sip provider it's not that it doesn't work, more than the only way to have some sound is to use the 'm' flag of the Dial() command to have the moh played during the ringing. Given that, i got some sound when the call is established ... -- Benoit ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Storing voicemail on samba share
It would probably be wiser to run an IMAP server and do imap storage instead of writing to a cifs-mounted directory... or use ODBC storage... assuming they are running a database server somewhere. I don't have any experience with having * write voicemail files to CIFS/SMBFS, but I also think its not something I would try... I've personally always found that most network file systems don't tend to handle disconnects (server reboots, network outages, etc.) very well. Mind you, it might of come along way since the last time I tried. -- Matt From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of OCG Technical Support Sent: Tuesday, May 06, 2008 3:34 PM To: 'Asterisk Users List' Subject: [asterisk-users] Storing voicemail on samba share A client has asked that our asterisk installation leverage their large investment in their existing data center infrastructure. We're thinking about putting the voicemail messages onto a Samba share (on their file servers). Any pros/cons to this? Does network/samba latency create choppiness? Thanks, MD ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
On Tue, May 6, 2008 at 1:38 AM, Benoit Plessis [EMAIL PROTECTED] wrote: We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 and it's quite unstable. We have ~30 IAX2 SoftPhones and encounter some Avoiding IAX destroy deadlock and now that we have added a Queue, it's worse than ever. The queue goes stuck quite often (agent are stuck in 'In use' state and if they logoff they can't log-in till an asterisk restart). There's an IAX issue with the security patch for 1.4.18.1... and 1.4.19.1. There's another thread on this. - Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how do I set callerid for incoming iax?
Tilghman Lesher wrote:
On Tuesday 06 May 2008 12:45:42 sean darcy wrote:
Using 1.6-rc8.
In iax.conf on the calling box, I have:
[iax-out]
.
callerid = sean 447
I even also put the same on called box.
But I can't seem to set the callerid:
exten =_NXX,1,Answer()
exten =_NXX,n,NoOp(${CALLERID(num)})
Answer(IAX2/iax-in-7, ) in new stack
NoOp(IAX2/iax-in-7, ) in new stack
So how do I set callerid?
iax-out != iax-in
So??
On the calling box,
[iax-out]
type=friend
username=iax-in
secret=password
peercontext=longdistance ; which also does extensions
host=
qualify=yes
trunk=yes
callerid = sean 447
On the called - receiving - box:
[iax-in]
type=friend
username=iax-in
secret=password
context=longdistance
qualify=yes
trunk=yes
callerid = sean 447
and then the cli shows:
-- Executing [EMAIL PROTECTED]:1] Answer(IAX2/iax-in-1, ) in
new stack
-- Executing [EMAIL PROTECTED]:2] NoOp(IAX2/iax-in-1, )
in new stack
I must be missing something. The name of the iax.conf context matters
somehow?
Thanks for any help.
sean
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Re: [asterisk-users] Storing voicemail on samba share
On Tue, May 06, 2008 at 03:33:31PM -0400, OCG Technical Support wrote: A client has asked that our asterisk installation leverage their large investment in their existing data center infrastructure. Weâre thinking about putting the voicemail messages onto a Samba share (on their file servers). Any pros/cons to this? Does network/samba latency create choppiness? I think a lot of that would depend on *how* you put the files there. smbclient? cifs mount? I would say write them locally, and then use a looper script to push them to the other server... though on reflection, Asterisk might not be happy with that. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
Are you saying the * server does NOT TRY to re-establish the BD connection ? Does your whole * SERVER freeze ? If NOT, what happens to you CDR records ? Anthony Francis wrote: Tilghman Lesher wrote: On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote: 5 maj 2008 kl. 19.58 skrev Tilghman Lesher: On Monday 05 May 2008 11:24, Johansson Olle E wrote: 5 maj 2008 kl. 17.51 skrev Tilghman Lesher: On Monday 05 May 2008 09:45, Johansson Olle E wrote: Another issue that we need to fix with the MYSQL driver is that we're lacking a connection pool. Everything seems to be handled over one connection to Mysql, which causes issues. That's not true. The MYSQL app generally uses multiple connections, one for each channel. The only way one might use only a single connection is by using a global variable to store a single connection id, but that method is not documented anywhere, AFAIK. You talk about the Mysql APP, but is this the case with the Realtime driver as well? No, the native Realtime driver uses a single connection. The ODBC Realtime driver generally uses a single connection but can be configured to use a separate connection for each query. So, we're back to where we started. A developer that can help us with a connection pool or a separate connection for each query would be a Nice Thing (TM). What issues are you specifically seeing that merit using multiple connections? I can specify an issue that would merit multiple connections, if the link to your db goes away Asterisk likes to freeze writing CDRs. I have a few remote * servers that this happens to. My solution so far has been to record CDR's to a local DB and then have a perl script that attempts to move them over to my transaction DB. I would suggest this solution to anyone who depends on their CDR records. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Receptionist SNOM-360
Hi to all, I got an Asterisk with 2 BRI(7 pstn numbers and 4 concurrent calls) and 15 SIP extensions. The receptionist has a SNOM-360. How many SIP accounts would you configure on that phone? Only one would be enough? One SIP account, has a limit on concurrent calls? I saw that the SNOM-360 can handle up to eleven SIP accounts. Thanks -- .:FaberK:. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Storing voicemail on samba share
On Tue, May 6, 2008 at 5:19 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Tue, May 06, 2008 at 03:33:31PM -0400, OCG Technical Support wrote: A client has asked that our asterisk installation leverage their large investment in their existing data center infrastructure. We're thinking about putting the voicemail messages onto a Samba share (on their file servers). Any pros/cons to this? Does network/samba latency create choppiness? I think a lot of that would depend on *how* you put the files there. smbclient? cifs mount? I would say write them locally, and then use a looper script to push them to the other server... though on reflection, Asterisk might not be happy with that. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) How about using rsync or FTP along with the Samba share? Just a thought, no real though or testing involved. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Please - Asterisk MYSQL interface seems to be eating data
On Tue, May 6, 2008 at 11:42 AM, Anthony Francis [EMAIL PROTECTED] wrote: Tilghman Lesher wrote: On Tuesday 06 May 2008 02:16:47 Johansson Olle E wrote: 5 maj 2008 kl. 19.58 skrev Tilghman Lesher: On Monday 05 May 2008 11:24, Johansson Olle E wrote: 5 maj 2008 kl. 17.51 skrev Tilghman Lesher: On Monday 05 May 2008 09:45, Johansson Olle E wrote: Another issue that we need to fix with the MYSQL driver is that we're lacking a connection pool. Everything seems to be handled over one connection to Mysql, which causes issues. That's not true. The MYSQL app generally uses multiple connections, one for each channel. The only way one might use only a single connection is by using a global variable to store a single connection id, but that method is not documented anywhere, AFAIK. You talk about the Mysql APP, but is this the case with the Realtime driver as well? No, the native Realtime driver uses a single connection. The ODBC Realtime driver generally uses a single connection but can be configured to use a separate connection for each query. So, we're back to where we started. A developer that can help us with a connection pool or a separate connection for each query would be a Nice Thing (TM). What issues are you specifically seeing that merit using multiple connections? I can specify an issue that would merit multiple connections, if the link to your db goes away Asterisk likes to freeze writing CDRs. I have a few remote * servers that this happens to. My solution so far has been to record CDR's to a local DB and then have a perl script that attempts to move them over to my transaction DB. I would suggest this solution to anyone who depends on their CDR records. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP I would not run MySQL on the local box. I would simple use Asterisk's csv CDRs and then use some script to import the CSVs into a database residing on another server using some sort of script. Depending on your needs, you could probably run that during low call volume. I also think that you adapt the free queue_log to database script by Queuemetrics to do what you want on the fly. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Melbourne Asterisk night
Tomorrow night is the monthly Asterisk night...in melbourne (australia)... The usual stuff - get together, eat, show off tech toys. At the Pint on Punt, from 7pm. later, PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] better enumlookup handler
Does anyone have a better ENUM lookup handler than the built-in ENUMLOOKUP() function? The built-in function does not properly handle multiple return values such as: 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 u E2U+SIP !^\\+1866(.*)$!sip:[EMAIL PROTECTED] . 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 u E2U+SIP !^\\+1866(.*)$!sip:[EMAIL PROTECTED] . And thus does not handle roll-over should one be unavailable for whatever reason. There is this voip-info.org wiki page: http://www.voip-info.org/tiki-index.php?page=Asterisk+and+multiple+ENUM +entries but the downloads that it's pointing to seem to be dead. Sure I could take to writing an AGI script and probably be done it in a few hours, but why re-invent the wheel? Thanx, b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] phone status question
Hi, A makes call to B. B has connection problem with the server (say, the lan cable is unplugged). 1: A --- server 2: A --- server 3: server B In 2, server will send the ring to A and it will hear ringing tone. In 3, server will try to connect B until timeout. My question is: A will still wait for B but B is physical unreachable. Can I set the number of retry time in server to try to reach the destination instead of the timeout? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie alert: VoIP hardware
Hello, Please forgive me for i'm not an asterisk user yet. I've done as much research as I can .. and have the following questions. I'm setting up a new office and a home office and i'm shopping for hardware. Office: 2 analog lines Hardware: TDM412B (2 FXO, 1FXO) Link: http://www.voipsupply.com/index.php?cPath=99_555_556 Cost: $303 Home: 1 analog line Hardware: TDM421B (2 FXS, 1 FXO) Link: http://www.voipsupply.com/product_info.php?products_id=3980 Cost: $300 Questions: [1] Can I use oslec for echo cancellation? I'll have beefy hardware. Is echo cancellation necessary? [2] Can I get PCI express x1 cards for the same price? I'm on budget, Any other cards (sangoma? rhino?) that might work well? I'm sure these questions have been asked before.. :-) Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue
hi, all is there any way in queue app. to execute asterisk app. after Queue() app. i.e [myplan] exten = _X.,1,Answer exten = _X.,n,Queue(myqueue) exten = _X.,n,Background(file-to-play) exten = 1,1,Playback(thnks) exten = 2,2,Playback(by) Is these possible above situation , how thnks, Bhrugu Mehta ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
