Re: [asterisk-users] syntax
Your sound file needs to be in the asterisk sounds directory. Another thing is that you may not have to put the file extension in the name if the file is in the proper place as well. Try that and see what happens. Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron Sent: Sunday, February 07, 2010 7:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] syntax I am trying to understand .call files. The logs seems to indicate issues with the audio file that I am trying to have played when the call is connected. Any thoughts? Some sample files and logs to console are shown. zipp-code.call Channel: SIP/callwithus/12023519259 Application: Playback Data: zipp-code.gsm [r...@localhost tmp]# touch zipp-code.call [r...@localhost tmp]# vi zipp-code.call [r...@localhost tmp]# mv zipp-code.call /var/spool/asterisk/outgoing/ -- Attempting call on SIP/callwithus/12023519259 for application Playback(zipp-code.gsm) (Retry 1) == Using SIP RTP CoS mark 5 [Feb 7 18:44:07] WARNING[20197]: file.c:635 ast_openstream_full: File zipp-code.gsm does not exist in any format [Feb 7 18:44:07] WARNING[20197]: file.c:936 ast_streamfile: Unable to open zipp-code.gsm (format 0x2 (gsm)): No such file or directory [Feb 7 18:44:07] WARNING[20197]: app_playback.c:447 playback_exec: ast_streamfile failed on SIP/callwithus-03d98080 for zipp-code.gsm [Feb 7 18:44:07] NOTICE[20197]: pbx_spool.c:357 attempt_thread: Call completed to SIP/callwithus/12023519259 -- Attempting call on SIP/callwithus/12023519259 for application Playback(yvrspecialemail) (Retry 1) == Using SIP RTP CoS mark 5 [Feb 7 18:54:58] WARNING[20228]: file.c:635 ast_openstream_full: File yvrspecialemail does not exist in any format [Feb 7 18:54:58] WARNING[20228]: file.c:936 ast_streamfile: Unable to open yvrspecialemail (format 0x2 (gsm)): No such file or directory [Feb 7 18:54:58] WARNING[20228]: app_playback.c:447 playback_exec: ast_streamfile failed on SIP/callwithus-03d98080 for yvrspecialemail [Feb 7 18:54:58] NOTICE[20228]: pbx_spool.c:357 attempt_thread: Call completed to SIP/callwithus/12023519259 -- _ -- 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] Help with concurrent VoIP calls
If you've got a bellsouth dsl connection because of the way their system works even with doing qos on the link you can really only do about 8 calls before you start to run into problems with their setup. Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Timms Sent: Saturday, November 07, 2009 2:45 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Help with concurrent VoIP calls Hi. I'm having trouble figuring out why I'm not able to make many concurrent VoIP calls on my system. I'm not aiming for a huge number, because I have purposely bought a low powered system, but I would think that I could get more. Here are the details: I have a small-form-factor Asterisk server with an Intel Atom 230 CPU (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu Server 9.04 with the default Debian package manager installation of Asterisk. (version 1.4) Here is what is going on: I'm making outgoing calls (with .call files) via SIP (using Vitelity's service, if anyone wants to know) with about 55.0 ms latency between my Bellsouth DSL connection their servers. I'm using GSM-format prompts with GSM encoding (disallow=all, allow=gsm in sip.conf) and I'm able to make about 7 concurrent calls. I have a very fast internet connection, so there is still plenty of bandwidth, and the top command shows that Asterisk is only at about 5% CPU and 10% RAM. Even with only 7 calls, a landline phone will skip occcasionally, but cell phones have perfect quality. I don't think that 7 calls is very many, I'll be happy if I can get 10 good-sounding calls. Can anyone give suggestions? (If this has been hashed out elsewhere, I'm happy with a link to more information!) Thanks. ___ -- 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] Upgrading Asterisk in Trixbox installation
One thing I kind of like that Trixbox does is their endpoint manager. That's about the only feature I haven't been able to replace. Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Monday, August 31, 2009 4:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Upgrading Asterisk in Trixbox installation On Mon, 31 Aug 2009, ilker Aktuna wrote: Thank you. That was quick and helpful :) Then I'll just make and make install What should I backup, in case of rollback requirement ? That's a bit tougher. At the least /usr/lib/asterisk/modules, /etc/asterisk, and /usr/sbin/asterisk... someone else may need to chime in here... I've always been a fan of trixbox, and I have done a lot of installations, but when it comes down to it all I really want it for is for a quick installations of asterisk and FreePBX. I don't think I actually use any of the trixbox-only features. I've also been enamored with Ubuntu of late, and have dumped CentOS. YMMV, but you might consider starting over with a clean build of the linux of your choice, and doing asterisk + addons + FreePBX from source. j Thanks. - Original Message - From: Jeff LaCoursiere j...@jeff.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 31, 2009 11:15 PM Subject: Re: [asterisk-users] Upgrading Asterisk in Trixbox installation On Mon, 31 Aug 2009, ilker Aktuna wrote: Hi, My Trixbox 2.8.0.1 installation includes the following Asterik version: 1.6.0.9-samy-r27 I am having some problems with it and I think they might be solved if I use the latest Asterisk version. Is it a good idea to update Asterisk in Trixbox externally ? I've done it in the 1.4 branch. Is it safe ? Should be, as long as you stay within the same branch. That being the case, I would stick with 1.6.0.14 if I were you. Make sure you don't make samples :) j If so, which version should I prefer ? 1.6.1.5 or 1.6.0.14 ? Thanks, ilker ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording Calls
if you are running Asterisk in front of the other pbx you can record the calls that you send to the other system. You will either need some type of pri interface to connect between the two systems if digital and some fxs interfaces if analog. Tom _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mr. Rodriguez Sent: Wednesday, July 29, 2009 3:14 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Recording Calls Greetings to all. this is my first question, and but that nothing is for consulting if this with asterisk can be realised. I have a commutator 3com, connected to 20 telephones of the same mark, my necessity right now is to be able to record the calls that enter the commutator, and wanted to know if this is possible with asterisk. by its attentions, thank you very much ___ Carlos Rodriguez Torreon Coahuila _ Messenger cumple 10 años de ser parte de tu vida ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Kamailio NAT problem
Try putting nat=yes in your asterisk peer Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joao Gomes Pereira Sent: Monday, July 27, 2009 9:50 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and Kamailio NAT problem Hello Im using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is behind NAT. X-Lite and SNOM phones behind NAT work fine. But when I try to connect with an Asterisk behind NAT, the Asterisk client doesn't receive sound. I already tried in 2 different NATs, with no firewalls. This is my Asterisk config: [kamailio] type=peer host=xxx.xxx.xxx.xxx disallow=all allow=ulaw allow=alaw allow=gsm allow=g726 qualify=1000 username=my_username fromuser=my_username secret=password sip*CLI sip show peers Name/username HostDyn Nat ACL Port Status kamailio/my_username xxx.xxx.xxx.xxx 5060 OK (890 ms) Is there something missing in my SIP.CONF to improve the compatibility with Kamailio? How can I debug the RTP stream in Asterisk? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt ___ -- 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] Configuring Deltacom pri in Florida
Hi guys, Trying to do a pri pass-through with a duel port t1 card from Digium. I have port 1 setup to come in and out from Deltacom and port 2 to feed the pbx. Incoming calls work from Deltacom just fine, but outgoing do not. Anyone know what switch type I may need to get the dial-outs to work or anything else I might need? Thanks, Tom ___ -- 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] Is it possible to do this? (forward a call w/3-way calling)?
It may be possible. There is a flash application in Asterisk. Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Piszcz Sent: Saturday, June 13, 2009 9:59 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Is it possible to do this? (forward a call w/3-way calling)? Hello, I have regular phone service (not VoIP) with an SPA3102. It works fine, I can dial out, incoming calls work as well, no issues. With the regular phone service, while I am on a call, I can initiate a 3-way call via: 1. Press [FLASH] 2. Dial the number. 3. Press [FLASH] Would this be possible via asterisk to 'forward' -- not in the example below but essentially to open a 3-way call (dial to another number?) http://www.voip-info.org/wiki/view/Asterisk+cmd+Flash Justin. ___ -- 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] VoIP over satellite internet
Voip over satellite can be done if enough bandwidth is reserved properly for it. Use the g729 codec and ask for 24 kilobits of upstream cir and you should be fine. Also you'll want to mark your packets with the EF tos bit in sip.conf. If done right the delay isn't too bad. Yes you can tell it is there, but as long as your latency doesn't fluctuate too much you should be alright. From my experience as far as satellite platforms are concerned go with Idirect when possible. I've also used the Viasat systems as well and from all the installs I've seen the hub operators never really understood how to run the hub to optimize for voip so I've pretty much given up on that platform. It seems that Idirect operators are the only ones who know how to make the system work well. If you want the best of the best nothing touches C-band though. Yes its low bandwidth, but C-band you can bet your life on... Most low end sat services are way over subscribed to start with and when you add voip in to the mix voip is the first thing to lose. Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Fort Sent: Friday, May 08, 2009 11:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] VoIP over satellite internet Could those on the list who have used or tried to use VoIP over a satellite internet connection comment on how well it works or if it even works at all in a reliable way. What is the effect of latency on the VoIP path and how much is generally tolerable? routing via satellite adds about a quarter second of latency to the path. Is that too much? Eric ___ -- 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] Canreinvite=yes // native bridging // 2 sip channels with different Call-ID
Asterisk still controls the signalling, but the audio path should be going through the phones directly. Fire up a tcpdump on the Asterisk server to varify this. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens Sent: Saturday, April 18, 2009 5:33 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Canreinvite=yes // native bridging // 2 sip channels with different Call-ID I have 2 SIP-clients defined in my sip.conf : [GXP1200] type=friend context=intern host=dynamic username=GXP1200 secret=testpaswoord canreinvite=yes [BT201] type=friend context=intern host=dynamic username=BT201 secret=testpaswoord canreinvite=yes When I make a call from one to another this is displayed on the CLI : -- Executing [...@intern:1] Dial(SIP/GXP1200-093900c8, SIP/BT201|30) in new stack -- Called BT201 -- SIP/BT201-09395070 is ringing -- SIP/BT201-09395070 answered SIP/GXP1200-093900c8 -- Native bridging SIP/GXP1200-093900c8 and SIP/BT201-09395070 From voip-info.org I understand that 'canreinvite' means that the SIP-client will re-invite the other client, so that Asterisk is no longer in the path... This is indicated on the CLI with 'native bridging'. Then why are there 2 sip-channels with a different Call-ID ? The output shows that Asterisk is still in between ! asterisk*CLI sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 192.168.x.x GXP2020 4684b544470 00103/0 0x4 (ulaw) No Tx: ACK 192.168.x.x BT201 1212e00ffa1 00102/43234 0x4 (ulaw) No Tx: ACK 2 active SIP channels Is there something that I misunderstand here ?? Thanks for the feedback on this ! Greetingz, Jonas. ___ -- 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] codec payload size
I believe 20 is the standard. 30 is where it might be tricky. Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, March 31, 2009 1:37 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] codec payload size I am about to connect to a new provider who requires 20ms payload sizes in g729a. Is this configurable on asterisk? Is 20ms the default? Cheers, j ___ -- 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 on EC2 cloud computing - priceassumptions - your brain needed
You'll need to use sip or some other network based protocol to provide access to the pstn. These boxes are virtual machines and you don't have any kind of access to the physical hardware on the machine itself. tom _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Sunday, February 15, 2009 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk on EC2 cloud computing - priceassumptions - your brain needed How do provide PSTN access to such hosted boxes ? ___ -- 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 Test Lab
The cable needed for this is a different cable than an ethernet cross over. I have actually done this same thing today with a Samsung 100 system and Asterisk 1.4.20.1 and Zaptel 1.4.11 and things work great. A question of my own: I know I can emulate the network side of a pri connection, but can I do this same trick with other t1 standards like ani and others? If I can be a client on the different t1 types, does this also mean I can be the server side and feed back the different standards to legacy equipment as well or are there some limitations to this? Thanks, Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Wilson Sent: Friday, February 13, 2009 4:19 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] PRI Test Lab Hey Everyone, I would like to start testing/playing with PRI channels but I don't have access to a PRI line. Is it possible to do the equivilent of a crossover between two PRI Cards (say Digium's TE120P)? What I was thinking is that I could set one asterisk box up with a PRI card set as the TE and provide clocking and another box exactly the same but with the card setup as NT. I think I would also need to wire up the correct type of crossover as a standard ethernet crossover would not work or would it? Thanks in advance. Lee ___ -- 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] NAT router for Linux
Hi, Are you having problems with sip calls or just using Gtalk? If you are behind a nat router you may need to forward in to your server port 5252. Check out the /etc/asterisk/gtalk.conf and /etc/asterisk/jabber.conf files. Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julien Claassen Sent: Saturday, January 24, 2009 12:32 PM To: asterisk users mailinglist Subject: [asterisk-users] NAT router for Linux Hello everyone! This is my problem: I try to do gtalk, but my asterisk server uses the local IP 127.0.0.1 or perhaps the 192.168.*.*. Now I've heard, that a NAT router can help there. I was told it's the way the windows-world does the trick, when they sit behind a router/phonebox/modem. Does anyone know a good software that will do the trick on Linux? I'm running Debian Lenny and one important thing: I can't use a GUI to configure anything. Any help is highly apreciated! Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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] PAP2T provisioning
I'm not sure if this trick will work with this device, but I was able to pull down a spa8000's config by connecting to: http://ipaddress/admin/spacfg.xml Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, January 20, 2009 6:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] PAP2T provisioning Anyone have an example XML file for the PAP2T? Cheers, j ___ -- 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] Configuring Linksys spa8000 devices through xml
I was able to send the xml directly to the device like: http://ip/admin/resync?tftp://ip_of_tftp_server/myconfig.xml This is a great start to what I want to do, but isn't really the end goal. I put a call in to Linksys about this so maybe they'll call me back today. Another issue I've got with these devices is on about 1 out of every 3 devices I get a high pitch noise on the line when I go to place a call so their hardware QA seems to be lacking in some way. Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies Sent: Monday, January 12, 2009 7:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Configuring Linksys spa8000 devices through xml I did this a long time ago, and just based it on a PAP2T XML configuration, with 8 lines instead of 2, and it worked fine. Sorry I don't have any useful examples to hand anymore. Are you sure it is not just a missing slash or angle-bracket in your source XML? Try opening it in a browser to see if it parses cleanly. Regards, Steve 2009/1/11 Tom Moore tommym2...@gmail.com: Hi guys, Anyone have experience using the Linksys spc tool to provision devices? I've been fighting with it all afternoon and can not get the tool to parse a xml file properly. I can get it to generate me a .cfg file from a text file, but when I tell the unit to pull down the cfg file it doesn't really look like any of my settings were even set from the file I generated. Thanks for any help you may be able to offer, Tom ___ -- 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] Configuring Linksys spa8000 devices through xml
Hi guys, Anyone have experience using the Linksys spc tool to provision devices? I've been fighting with it all afternoon and can not get the tool to parse a xml file properly. I can get it to generate me a .cfg file from a text file, but when I tell the unit to pull down the cfg file it doesn't really look like any of my settings were even set from the file I generated. Thanks for any help you may be able to offer, Tom ___ -- 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] Log level of 500 Server Internal Error.
Hi, I've started noticing these messages today myself specifically with Broadvox. Are you using this carrier or someone else? Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins Sent: Friday, November 21, 2008 12:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Log level of 500 Server Internal Error. Hi, VERBOSE[6120] logger.c: -- Got SIP response 500 Server Internal Error I just noticed that i sometimes get those back from provider. They are currently general SIP message log entries with verbose level 3. I wonder if such SIP fails could generate at least WARNING in log? Currently i'm checking logs for warnings and errors, so i probably have missed those.. It would be great indication that something is not ok - either outgoing trunk or local phone is bad. Any opinions? Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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 No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.175 / Virus Database: 270.9.9/1803 - Release Date: 11/21/2008 9:37 AM ___ -- 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] Dedicated Servers
Right now I'm currently working with Sagonet www.sagonet.com. They've been good for me for a good while now on servers and uptime. Tom _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sahil Gupta Sent: Friday, November 14, 2008 3:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dedicated Servers Hi, I am looking for a reliable provider that can provide 3 dedicated linux servers asap. Unfortunately, the provider I have used for YEARS has become way too slack in recent times and we have to move on. Cheers, Sahil No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.175 / Virus Database: 270.9.3/1786 - Release Date: 11/13/2008 6:01 PM ___ -- 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] Aastra phones and dns srv records
Never really tried just a dns entry. I'm wanting to give the phone the ability to sink up to which ever of my three servers are online. I've got all my Linksys gear doing this. Just wondering if I can get Aastra equipment to do this too. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Trevor Peirce Sent: Tuesday, October 14, 2008 1:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Aastra phones and dns srv records Tom Moore wrote: Hi guys, Does the Aastra line of phones work with dns srv records? I'm trying to get my 8133i to do this and in the settings it asks for ip addresses of registration and proxy servers. Does this mean that it will not just let me put the domain name in like other devices I have and then do fail over to other servers when needed? If these phones do not what phones do? Have you tried putting in a domain name to see what happens? I use domain names on all my Aastra phones. I don't think they have SRV support, bu they certainly do have DNS support. ___ -- 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 No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.173 / Virus Database: 270.8.0/1722 - Release Date: 10/13/2008 6:42 PM ___ -- 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] Aastra phones and dns srv records
Hi guys, Does the Aastra line of phones work with dns srv records? I'm trying to get my 8133i to do this and in the settings it asks for ip addresses of registration and proxy servers. Does this mean that it will not just let me put the domain name in like other devices I have and then do fail over to other servers when needed? If these phones do not what phones do? Thanks, Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
I will second this opinion. This may be going a little off topic, but is there a way to lock the voip section of the ata so that the end user can not change settings in this area, but as far as the ip settings go and other sections the user will be able to access them? Thanks, Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andres Sent: Monday, September 29, 2008 9:58 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ATA for large networks In other words, I'd really appreciate feedback from voip administrators (not from resellers) who have had experience testing their devices and are happy with them. I would recommend the Linksys SPA8000 (8 port ATA). It is as solid and reliable as the SPA2102. Andres http://www.neuroredes.com Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.169 / Virus Database: 270.7.5/1696 - Release Date: 9/29/2008 7:40 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Terrible Experience Net2phone A-Z termination
I've been happy with Broadvox if you can meet their minimum requirements. I use them as one of my termination carriers for both A-Z and domestic traffic and have been happy with their quality. their rates aren't the lowest for A-Z but good quality just the same. Anyone else have a review of Broadvox to share either good or bad? Tom _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of broadband Voice Sent: Thursday, September 25, 2008 7:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Terrible Experience Net2phone A-Z termination Who are you using now? We need someone that has international traffic with good rates and good quality. On Thu, Sep 25, 2008 at 7:32 AM, Igor Hernandez [EMAIL PROTECTED] wrote: Bruno Castelo Branco wrote: you can try inphonex.com http://inphonex.com/ Steve Totaro wrote: Try Bandwidth.com or Junction Networks. You get what you pay for. If you want a lower end provider, go with Vitelity, Gafachi, or even VoicePulse. I am not saying they are lower end on service necessarily, but on reputation and corporate image. Vitelity tested very well in a very limited time frame. VoicePulse was great too but they kept making changes that resulted in outages, if engineered properly, there should be no outage short of an act of God. Thanks, Steve Totaro On Thu, Sep 25, 2008 at 3:31 AM, broadband Voice [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I'm using Net2phone termination and the experience has been horrible for the past 2 weeks, I have put in several tickets and nothing has been done. I get a lot of congestion, channel unavailable and calls not going through. Does anyone use them? I have been using SIP debug to try to resolve it but to no avail. Are there any tier A-Z termination partners out there, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ 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 http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ 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 http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Funny Net2Phone comes up. We talked to them when we were starting out and they wanted to charge $500 setup fee because we had no volume. The guy said We have to charge this because we had many people coming to us without volume, so we charge this setup fee in order to allow us to still provide them service. Like that makes any sense to anyone. Either way, they had the worst rates in the market and claimed extremely high quality. I'm glad we didn't go with them. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com http://www.escapetel.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting two asterisks via IAX
Easy.
Just create a peer in each office that connects to the other, basic example
on server 1.
iax.conf
[office2]
type=friend
host=office2
disallow=all
allow=ulaw
context=internal_office_dialing
username=office1
secret=mypassword
trunk=yes
Create a peer on the office2 server to point back at office1 in the same
way.
extensions.conf on office1.
exten = _2XX,1,Dial(IAX2/office2/${EXTEN})
This is a simple setup where office1 has the 1XX extension block and office2
has the 2XX block.
Tom
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nuno Marques
Sent: Friday, August 29, 2008 11:36 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Connecting two asterisks via IAX
Hi,
I need to connect 2 asterisks in 2 different countries (A and B) for one
company so it's possible to make connections between the 2 offices.
For connectivity reasons (NAT traversal) i want to connect the 2 asterisk
with IAX so that when a user on office A connects via SIP to user on office
B the call is going trought IAX channel.
Can anyone give me an ideia how to accomplish this?
the schema:
(via SIP)(via IAX)
(via SIP)
Office A - Asterisk A --- Asterisk B
- Office B
Thanks in advance,
Nuno
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pri to sip interfaces
Most likely at this point what I should be using for a hardware platform that will work in a small area that I can put in place and leave it put and running for years on end like you can with other pbx equipment. The pbx installer I worked with says that 16 channel sip cards for the system cost around $1300 or so. I am also not sure that all the pbx's he has out in the field would be compatible with this sip card so I'm looking for hardware suggestions at this point I think. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, August 27, 2008 9:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Pri to sip interfaces Are you looking for a hardware suggestion or a software suggestion? PaulH Tom Moore wrote: No, these are mainly Samsung pbx systems. I know I can use Asterisk to do this but what be a solid platform to go with that can go in the phone closet? tom *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Darren Sessions *Sent:* Wednesday, August 27, 2008 9:22 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Pri to sip interfaces Are you using an Asterisk PBX? _ Darren Sessions [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 27, 2008, at 7:06 PM, Tom Moore wrote: Hi guys, What are your suggestions to people who have pbx systems that interface with the world over pri and want to convert them to sip interfaces so that they can use sip trunking? Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reliable wireless SIP phones
I have a client testing one of these and he is happy with it so far. I don't know if there are any known problems yet with this phone, but would be interested in knowing about your review. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fred Posner Sent: Thursday, August 28, 2008 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Reliable wireless SIP phones On Thu, 28 Aug 2008, Jaap Winius wrote: Hi list, Are there any reliable wireless SIP phones available on the market yet? My Linksys WIP330 should arrive today. I've always wanted to test how well it would work in hotspots... will let you know. Fred Posner [EMAIL PROTECTED] Tel: +1 (212) 937-7844 x501 Fax: +1 (954) 252-4187 www.teamforrest.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pri to sip interfaces
Hi guys, What are your suggestions to people who have pbx systems that interface with the world over pri and want to convert them to sip interfaces so that they can use sip trunking? Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pri to sip interfaces
No, these are mainly Samsung pbx systems. I know I can use Asterisk to do this but what be a solid platform to go with that can go in the phone closet? tom _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Sessions Sent: Wednesday, August 27, 2008 9:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Pri to sip interfaces Are you using an Asterisk PBX? _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Aug 27, 2008, at 7:06 PM, Tom Moore wrote: Hi guys, What are your suggestions to people who have pbx systems that interface with the world over pri and want to convert them to sip interfaces so that they can use sip trunking? Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Semi-OT Satellite?
Hi, using Asterisk over satellite can be done. Not all satellite providers are created equal and some are better than others. If you are going to do communications between offices that are connected over satellite office to office you may have a problem. My personal choice for satellite connections is the Idirect platform. Tom From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken Williams Sent: Saturday, August 23, 2008 9:44 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Semi-OT Satellite? We're entertaining moving our intranet to Hughes satelite for our remote locations. I'm curious if anyone with Asterisk servers has used satellite, and if so, is the latency an issue. My understanding is that you immediately introduce 250ms latency for travel time up and back down, however it is a much more direct connection then offered by traditional land lines. Perhaps someone has some other suggestions? We've started looking into Global Crossing as an alternative to have more control and reliability between all of our remote facilities, maybe this is a better alternative. Our biggest problem is most of our sites are in smaller cities where your bigger connections are more limited. Looking for any suggestions. Thanks, Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Global VoIP Calls?
I agree. You will probably get good ping times between your sites in Asia, but if your thinking about a back hall back to the states this is where your going to have the latency issues crop up. Tommy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Saturday, August 23, 2008 9:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Global VoIP Calls? On Sat, 23 Aug 2008, Gavin Henry wrote: Dear All, What setup would you recommend for making VoIP calls whilst bringing latency down between offices at: * Edinburgh * Kuala Lumpur * Singapore * Tokyo * Seoul * Beijing * San Francisco Some of the Asia offices are 300ms some 200ms. Any advice greatly apreciated. Probably not the right answer, but ... Find a local ITSP in each country and place all your outgoing calls via them and let them deal with it via the PSTN. Mayby not in the true spirit of VoIP, and not free either, but if it works and you get some good rates, then it might well be worth it. Or provide both solutions - let the offices call each other via VoIP, but if too laggy, fall-back to VoIP - PSTN... (- VoIP) You'll be at the mercy of your local Internet providers and usually, there's not a lot you can do to influence traffic routing - other than pick another ISP - maybe you can find out which ISPs use cable/fibre and which use satellite connections and favour the wired ones... But if you want to keep it VoIP, then what I'd do is get access to a PC in each location and start running traceroutes (use 'mtr' if you can) and work out the best paths - you might find that there are better ways then simply providing 6 IAX trunks at each location - eg. you might find it better to route calls from SF to Seoul via the Tokyo office (ie. use canreinvite=no to force the data path if using SIP or notransfer=yes in IAX with appropriate dial-rules) if SF to Tokyo to Seoul goes via cable, but SF to Seoul goes via satellite... So logon to those 7 asterisk boxes and run 6 mtr's from each to each other site - leave them going for an hour, then analyse the results. (Good luck!) However, you'll still be at the mercy of the ISPs who might change their routing on a day to day basis, depending on what their influences are... But at the end of the day ye canny change the laws o' physics! Gordon (Scottish, so fully licensed to utter that phrase ;-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Explain t38 and how it relates to Asterisk
Hi guys, What's the easiest way to explain to someone the state of t38 with Asterisk and what does / doesn't work in 1.4 stable from an end user point of view? I have a friend who uses an ata to connect to an Asterisk server and he asks me why he can not send faxes even though both end points support t38 and I have the canreinvite option set to yes in sip.conf I also have t38pt_udptl = yes Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto provisioning phones
Druid does I believe. Not sure about any others though. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Friday, August 01, 2008 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] auto provisioning phones Which Asterisk systems provide automatic provisioning of phones? Switchvox? ABE? The AA series appliances? Trixbox? I know that the VDEX-40 (Voiceroute) and Jazinga do this. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting up ring group
Hi guys, What's the best way to setup a ring group that contains 6 extensions so that when a call comes in there starts a 30 second timer and the first available device is rang instead of ringing all extensions at the same time? What I want it to do is cycle through the extensions and have the system ignore the ones that are busy and if there are not any free extensions in the ring group to have the system drop the caller to voicemail. If none of the extensions are present in the group I'd like to also drop to voicemail. Basically what I'm looking for is a multiple extensions version of the standard extension macro with multiple devices and the exten busy state ignored. Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up ring group
This works only half way. This gives the ring function I want, but doesn't take in to account the 30 sec timer to send to voicemail if the line is not answered. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ruddy G. Sent: Thursday, July 31, 2008 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Setting up ring group Why don't you just call the Dial application for each user, one after another ?? The ones that are busy will just go through. So, on the next priority, you dial another one. Tom Moore wrote: Hi guys, What's the best way to setup a ring group that contains 6 extensions so that when a call comes in there starts a 30 second timer and the first available device is rang instead of ringing all extensions at the same time? What I want it to do is cycle through the extensions and have the system ignore the ones that are busy and if there are not any free extensions in the ring group to have the system drop the caller to voicemail. If none of the extensions are present in the group I'd like to also drop to voicemail. Basically what I'm looking for is a multiple extensions version of the standard extension macro with multiple devices and the exten busy state ignored. Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up ring group
This is true. Probably is a hunt group. Different systems use different terminology for the same thing sometimes. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Komito Sent: Thursday, July 31, 2008 11:19 AM To: Ruddy G. Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Setting up ring group Sounds more like a hunt group than a ring group. Bruce Komito WPTI Telecom (775) 236-5815 On Thu, 31 Jul 2008, Ruddy G. wrote: Why don't you just call the Dial application for each user, one after another ?? The ones that are busy will just go through. So, on the next priority, you dial another one. Tom Moore wrote: Hi guys, What's the best way to setup a ring group that contains 6 extensions so that when a call comes in there starts a 30 second timer and the first available device is rang instead of ringing all extensions at the same time? What I want it to do is cycle through the extensions and have the system ignore the ones that are busy and if there are not any free extensions in the ring group to have the system drop the caller to voicemail. If none of the extensions are present in the group I'd like to also drop to voicemail. Basically what I'm looking for is a multiple extensions version of the standard extension macro with multiple devices and the exten busy state ignored. Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] issue with high latency
Not true. Voip is done over satellite every day and those ping times are at least 540 and upwards of in the 700's depending on the technology used. The key here is keeping the latency stable. If the packet flow fluctuates too much in latency this is when a problem arises. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Howes Sent: Tuesday, July 22, 2008 10:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] issue with high latency On 22 Jul 2008, at 14:36, Nhadie wrote: Pinging my.sipserver.com [202.203.204.205] with 32 bytes of data: Reply from 202.203.204.205: bytes=32 time=250ms TTL=56 Reply from 202.203.204.205: bytes=32 time=250ms TTL=56 Reply from 202.203.204.205: bytes=32 time=651ms TTL=56 Never going to work with that latency. I would say anything over 150 is probably pushing it. S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Interfacing pri card to legacy pbx
Hi guys, Can I use a Sangoma a101 to interface a legacy pbx to an Asterisk server? The pbx doesn't have sip and I want to come in off of a sip trunk and interface with the older system. I know I can use a pri card to hook in to the phone network, but can I use this same card to feed back the signaling as if I were the phone company to the older system? Thanks, Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interfacing pri card to legacy pbx
Actually what I'm doing is interfacing the legacy pbx and converting it to use sip for its way out to the world. The phone vender I'm working with says his system requires b8zs signaling and uses the esf frame type. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicolas Ross Sent: Tuesday, July 15, 2008 8:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Interfacing pri card to legacy pbx I cannot tell for sure for any system, but we have an old Portmaster PM3 hooked-up from one port of our Sangoma A104d card, another one being from telco. So, yes you can emulate the telco from a sangoma A10x card. Here's what I have in my zapata.conf : ;Sangoma A104 port 1 [slot:12 bus:0 span: 1] switchtype=national pridialplan=unknown signalling=pri_cpe group=1 channel = 1-23 ;Sangoma A104 port 2 [slot:12 bus:0 span: 2] echocancel=no pridialplan=national signalling=pri_net group=2 channel = 25-47 You might have noticed that the signalling is different for both port. pri_net being the telco emulatin one. The clock needs also to be set on master in the wancfg utility. Another thing, you might want to consider using a 2 port card for that, because the clock master needs a reference and I can't tell for sure if it'll work with a reference from another card. Regards, Nicolas Hi guys, Can I use a Sangoma a101 to interface a legacy pbx to an Asterisk server? The pbx doesn't have sip and I want to come in off of a sip trunk and interface with the older system. I know I can use a pri card to hook in to the phone network, but can I use this same card to feed back the signaling as if I were the phone company to the older system? Thanks, Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk dimensioning
How many calls do you expect to be going at one time? Do you have any sip trunks for the users to call out on? Unless this ratio really works for you I'm not sure a 15 to 1 ratio works for most people. I wouldn't just depend on a single server for this purpose. I'll leave it to the cluster guys to describe the ideal setup you should use. I have an idea of how I might do it, but I wouldn't want to get it wrong. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of voip crazy Sent: Wednesday, July 09, 2008 3:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk dimensioning Hello all, I need to install asterisk for 900 sip users with 2 PRI ports. It is posible to handle this number of calls/extensions with only one asterisk machine? Which is the best way to install that? two asterisk with openser. One asterisk with openser . Is it necesary run a SER server on this enviroment? Any clue will be welcomed. Thanks in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Default table layout for cdr logging with Mysql
Hi guys, I've been looking for a table layout that I should be using for cdr logging to a Mysql database. Everything I find seems to be a little different. What should I be starting with before I start adding custom fields in the future? Also note I want to import some Master.csv files in to this database as well. If I can't do this I'll just start fresh, but pack porting old cdrs would sure be nice. Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk peer definition with multiple host ip addresses
Hi guys, How would I configure a peer in sip.conf that has multiple host ip addresses? Can I just put multiple host=ip_address lines in the config file or will I have to create multiple peers each one with a different host ip address setting? A provider I am trying to hook up with gave me multiple ip addresses to accept calls from for incoming calls and multiple addresses that I can send calls too and I'm wondering what the best way to handle this is on my side? I will be working with Broadvox and they will be pushing calls to my server from a few different sources it looks like. Thanks, Tom ___ -- 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] Asymmetric codecs in IAX2 trunk
I don't think that you can have different codecs in different directions on a single call. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Magdy Salama Sent: Friday, May 02, 2008 9:58 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asymmetric codecs in IAX2 trunk I'm trying to implement the following setup: Internet ---RTP---Asterisk1IAX-trunk-over-VPNAstrisk2PSTN The outbound bandwidth at asterisk2 (PSTN) is limited and expensive, so I want to keep it low as possible. I want to use Speex VBR for the outbound stream and keep the inbound stream at G.711 in the IAX2 trunk. Speex streem will be transcoded to G.711 at Astreisk2. My question is: Does IAX2 trunk support asymmetric codecs (different codec at each direction)? regards, Magdy Salama ___ -- 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] FSX gateways
Hi guys, What are some reliable sip to FSX gateways with four ports and eight ports? I've used some Linksys and Grandstream devices and I find that at unexplained times there will be echo on the line. Sometimes this happens on the end where the devices is placed and sometimes this happens on the other end. Also are there devices that support codecs such as ilbc or gsm so that I can put four to eight phones on a dsl line? Thanks for suggestions. Tom ___ -- 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] SIP carrier billing technicalities
I'm not exactly sure how this works myself either. Most sip providers charge based on the destination you are calling. I'm not sure how providers do this if they have different rates for local area and national calling areas. I would assume that if you have say a 212 did if you set the caller id string to a 212 number and have a local calling area of 212 that since those both match the provider would give you the call for free. I could be wrong though. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert McNaught Sent: Monday, March 24, 2008 7:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP carrier billing technicalities Hi, Does anyone know anything about the following? In a hosted environment where several area DIDs are provisioned on a single server, how do most carriers establish the origination DID, number. Asterisk allows us to modify the CallerID, name, number and DNID channel variables before dialling out via SIP. Most carriers will allow us to spoof a callerID when placing a call, and pass it forward. We can also spoof the DNID also before the call is placed, although I am not sure this is carried forward to the outgoing proxy, using SIP. Do most carriers the carrier just use CallerID as an origination number? As far as I am aware, the concept of a BTN is gone with SIP Does this mean its possible to save money on your bill, by modifying CallerID so that the origination number is seen to be a local call? anyone care to comment? Robert ___ -- 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] Is Asterisk ready for Prime-Time?
I think what Bill is concerned about is if Asterisk is good enough for his operation. When you talk about Asterisk there are a number of ways to set it up. Some ways of setting it up involve a gui interface, some are pure text files and some are databased in how they handle the config data. As far as managing a voice system four to five hours of maintenance time for a 30 station system is entirely too much on a weekly basis. Asterisk most certainly can be a solution that can work for your situation. If deployed right it can handle on a single server much more than what you are asking it to do. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: Wednesday, March 19, 2008 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is Asterisk ready for Prime-Time? Bill Andersen wrote: This is not a troll. I've used my real email because I want this taken seriously. I'm not trying to make anyone mad, I just want some real discussion on this issue. Please bare with me... I'm a USER of Asterisk. We purchased 3 commercially available Asterisk Based PBXs a little over a year ago. (I won't mention which one at this point - I don't want to bad mouth them - yet!) Two of the systems are very small (5 SIP lines/6 Polycom phones). The third is on a PRI with 30 Polycom phones. My smaller sites work pretty good. I've only had to restart Asterisk every month or so. However, my 30 station system is a continuous headache. I average a restart at least once a week. Sometimes a couple of times in the week. I'm always being called to fix something that just stopped working. I DON'T WANT TO GET INTO A Well, don't just complain, tell us your setup and we can help you get it working. This list HAS helped me figure out some of the issues. THANK YOU! But the purpose of this post is more of a fact finding mission. 1) Was choosing Asterisk for our company the wrong decision... a) IF... I expect a phone system to just work. Once it is configured, a phone system should just work with very little attention. My previous system was a Comdial with external voice mail on a DOS based PC. I LITERALLY WENT OVER 4 YEARS WITHOUT HAVING TO REMOVE POWER TO THE COMDIAL CONTROL OR RE-BOOT THE VOICE MAIL PC. b) IF... I really only need a phone system that allows an operator to answer each call and transfer them to the appropriate person. I need voice mail, but very little auto attendant features (mostly after hours). All the bells and whistles that Asterisk offers are cool, but don't bring that much to the table for our purpose. c) IF... Stability is more of an issue than high end features? 2) Are there any users out there that really DO have an Asterisk system that just works like clockwork? I'm saying, once setup, run for a year (or more) without any issues? 3) If SO, Should I simply consider a different vendor? 4) If NOT, and if my expectations are that a system SHOULD just run and run without any problems. Is Asterisk simply not my solution. Is Asterisk not REALLY ready for production. Because in my mind (as a user of phone services), dealing with the phone system, even on a MONTHLY basis, means that the system is NOT really production ready... Before we installed an Asterisk based PBX, I spent maybe 4 hours per YEAR with phone issues (setting up a new station?). Since we moved to an Asterisk based PBX, I spend 4 hours (or more) every WEEK! Am I expecting too much? Bill I don't think you are expecting too much. We have:- 130 physical extensions including 24x7 inbound call centre Debian on Dell server [EMAIL PROTECTED]:~# uptime 13:15:31 up 192 days, 23:49, 2 users, load average: 0.00, 0.01, 0.00 (Power was removed to switch to new UPS) asterisk*CLI show version Asterisk 1.2.24 built by root @ asterisk on a i686 running Linux on 2007-09-08 17:17:07 UTC asterisk*CLI show uptime System uptime: 63 days, 4 hours, 26 minutes, 40 seconds (Asterisk was restarted after queue config changes) We had a single power supply and single drive fail in one incident in Feb 2007 (one drive of RAID 1). System stayed up but was taken down for 15 minutes to swap the drive. PS was hot-swapped when it arrived later. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] RedFone foneBRIDGE2 2e1 - anyone used it oranother TDMoE bridge?
Also to add to the last post does this device have hardware echo cancelation? if it does it could be a great replacement, if not may not be what I'd really want to use. Thanks, tom _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of arkda Sent: Thursday, March 13, 2008 3:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RedFone foneBRIDGE2 2e1 - anyone used it oranother TDMoE bridge? I've been asked to look at deploying Asterisk in a high availability environment and I've been looking so I've been searching for methods to decouple the voice PRI circuits from the Asterisk server so failover to another server could take place. I've been looking at the RedFone foneBRIDGE2 2e1 product here: http://www.mapleleaf-technologies.com/webstore/redfone_fonebridge2_2e1.php Has anyone used this device (or something similar)? What were your thoughts on it? On the surface this seems like a perfect method of building high availability Asterisk environments, but I'm a little hesitant to spend a few grand just to find out it's a pipe dream. ___ -- 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] I need the least expensive way to do this
Probably an 8 port gateway. This would give you four of each and a network jack to tie back to your asterisk server. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Timothy C Litwiller Sent: Saturday, March 01, 2008 2:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] I need the least expensive way to do this I never did see this get to the list. Tim Litwiller wrote: For my church school we need a way to connect 3 room phones, 1 office phone and 2 phone lines. so I need a device or several that i can connect to 2 pots phone lines and at least 3 plain old wall phones. I'll donate a sipura 941 for the office. What would be the best product to get 2 fxo ports and 3 fxs ports. ___ -- 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] Call limits per server with Iax
If I'm using an Iax trunk between two sites what is the suggested number of calls to pass over this trunk before I run in to problems? Also is this number based on a per peer basis or all Iax calls going through the server in general? I know this will depend on the bandwidth I have between the sites, but lets take this variable out of the question and assume I have more than enough bandwidth to go around for the calls I want to process. Also to make things easy lets say I have g729 encoder cards in each server at both ends of the link. Will Iax scale well enough to allow me to pass the 92 calls or whatever the number of g729 encoder cards have for the number of encoding licenses on the cards? Thanks, tom ___ -- 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] Zaptel and Asterisk compilation
You can install the packages you want with yum from any location on the system. You do not have to be sitting in /usr/src to do this. /tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Saturday, February 16, 2008 5:22 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Zaptel and Asterisk compilation Hi All; Are the following needed to be installed in the same location of Asterisk source and Zapata source, or it can be in any place? For example, I have asterisk at /usr/src/asterisk-1.4 and I have zaptel also at usr/src/zaptel-1.4, so where I have to instal the following: yum install bison yum install bison-devel yum install ncurses yum install ncurses-devel yum install zlib yum install zlib-devel yum install openssl yum install openssl-devel yum install gnutls-devel yum install gcc yum install gcc-c++ Regards Bilal Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- 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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.6/1280 - Release Date: 2/15/2008 9:00 AM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.6/1280 - Release Date: 2/15/2008 9:00 AM ___ -- 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 gsm bandwith calls
How much bandwidth does speex use usually? Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, September 28, 2007 11:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX gsm bandwith calls Andrew Joakimsen wrote: On 9/26/07, Tom Moore [EMAIL PROTECTED] wrote: If you've got a bandwidth of something that low you'll probably want to use g723.1 or g729 on this line. If your lucky you'll be able to place two calls at once over this link. You won't be able to do anything else though. Tom If you really want to maximize your bandwidth try LPC codec! You can probably squeeze 5 maybe 6 calls on there... and sound like a robot. Speex rocks! Thanks, Steve Typed using my fingers on my laptop in the Phoenix Airport waiting for my flight home from Astricon. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.488 / Virus Database: 269.13.33/1034 - Release Date: 9/27/2007 5:00 PM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.488 / Virus Database: 269.13.33/1034 - Release Date: 9/27/2007 5:00 PM ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 gsm bandwith calls
If you've got a bandwidth of something that low you'll probably want to use g723.1 or g729 on this line. If your lucky you'll be able to place two calls at once over this link. You won't be able to do anything else though. Tom _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dario Mendez Sent: Wednesday, September 26, 2007 12:10 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] IAX gsm bandwith calls Hi everybody I have 2 asterisk server connected by iax trunk using gsm over a 64Kbps Frame relay circuit, my questions are:whats is bandwith of each call?, and how to limit this on asterisk? Thanks.. No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.488 / Virus Database: 269.13.30/1030 - Release Date: 9/25/2007 8:02 AM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.488 / Virus Database: 269.13.30/1030 - Release Date: 9/25/2007 8:02 AM ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Sip Providers
I know that Voicepulse can do this. By default the offer 4 channels, but you can buy the other two or how many others you need as well. http://connect.voicepulse.com Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan Sent: Wednesday, July 18, 2007 5:07 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sip Providers Asterisk Users, I have Asterisk PBX System running at my work. The system is working great. Currently, I have Broadvoice as my sip provider and I am not completely satisfy with their service. Broadvoice only allows 2 simultaneous calls, which hinders my company's communications ability. I am looking for a sip provider that would work with Asterisk and allow at least 6 simultaneous calls, locally and internationally. Of course the voice quality, pricing, number portability are the main determining factors. I will have a T1 connection at the office, so bandwidth would not be an issue. Any thoughts on this matter would be greatly appreciated. Thanks. Best Regards, John _ Need a brain boost? Recharge with a stimulating game. Play now! http://club.live.com/home.aspx?icid=club_hotmailtextlink1 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.8/906 - Release Date: 7/17/2007 6:30 PM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.8/906 - Release Date: 7/17/2007 6:30 PM ___ --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 with iax2 over satellite
Hi guys, I'm in the process of setting up an Asterisk server over a satellite connection to allow people on a remote island to place and receive calls over the pstn. What are the ideal settings I should use in iax.conf for the optimal operation over satellite besides the normal options for the type=friend peer? Does anyone have this working? I an place calls as things are now, but there is a lot of drop out in the audio. I get a lot of receiving mini frame before full voice frame errors especially the first 5 or 10 seconds of the call. Thanks, Tom No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.2/894 - Release Date: 7/10/2007 5:44 PM ___ --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 with iax2 over satellite
Sip would probably work well in a single phone situation, but what I'm trying to do is use multiple phones over a single trunk connection. Using sip though do you have a few seconds at the beginning of each call where the audio is not clear? On our link when I tried a sip phone the connection was unstable for about 5 seconds before the two people could hear each other well enough to have a conversation. Also another things I noticed is that with either sip or Iax that when I call came in the latency shot through the roof up to 2.5 seconds or so for the first 30 seconds of the call before it settled back down to the usual 600 to 700. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: Sunday, July 15, 2007 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk with iax2 over satellite I have a client that is using SIP over satellite with G729, VAD and Jitter buffer. The calls are coming in great. - Original Message - From: Tom Moore [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Sent: Sunday, July 15, 2007 5:50 PM Subject: [asterisk-users] Asterisk with iax2 over satellite Hi guys, I'm in the process of setting up an Asterisk server over a satellite connection to allow people on a remote island to place and receive calls over the pstn. What are the ideal settings I should use in iax.conf for the optimal operation over satellite besides the normal options for the type=friend peer? Does anyone have this working? I an place calls as things are now, but there is a lot of drop out in the audio. I get a lot of receiving mini frame before full voice frame errors especially the first 5 or 10 seconds of the call. Thanks, Tom No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.2/894 - Release Date: 7/10/2007 5:44 PM ___ --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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.2/894 - Release Date: 7/10/2007 5:44 PM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.2/894 - Release Date: 7/10/2007 5:44 PM ___ --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
