Re: [asterisk-users] is DNS SRV enough for failover?
Hi Sir. This is the result of my query: ~$ host -t SRV _sip._udp.mydomain.com _sip._udp.mydomain.com has SRV record 0 1 5060 sip-1.mydomain.com. _sip._udp.mydomain.com has SRV record 0 3 5060 sip-2.mydomain.com. is that what you meant on having at least 2 SRV record? does this mean i need a UA capable of querying DNS SRV? i know it's not a real failover but at least the UA should still try to register on the other server if it cannot connect. Those records look fine. Priority is 0 for both, which is the highest, and the higher one with weight of 3 should be selected most of the times. My guess is your UA is either not doing SRV queries or is not following them in any case. To know if it is actually doing the queries try to sniff the traffic with Wireshark and analyze it. Andres http://www.neuroredes.com thank you regards, nhadie Andres http://www.neuroredes.com TIA regards nhadie ___ -- 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] Software patents (was G723 on asterisk 1.4.1)
On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote: On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: It is completely illegal in any country that recognizes patents. You mean countries that recognize software patents, right? As resident of country where the file is hosted - yes we don't have software patents, they have been proposed to EU and reject few years ago. So by law - software is algorithm and can't be patented. In local laws we even are allowed to reverse-engineer software for needs of compatibility and interoperability. So, writing code for commercial codec and using it for interoperability with hardware devices (you purchased) is allowed by law. Damn, we even have a law that don't allow bittorrent trackers, as bittorrent file is considered breaking copyright law.. Ironic :p Please do NOT discuss ways to use unlicensed codecs on this list or any other forum provided by Digium. This has been discussed multiple times as to why not, and I don't feel like rehashing the argument again. I did not know you were a moderator on this list. contributory infringement What if I make a page that explains the patent issues and then provide a link to http://asterisk.hosting.lv/ from that site and only provide people on this list a link to my site? What if I provide a link to the Google search for asterisk g723? Where do we draw the line? If that site is so illegal, why hasn't it been taken down? Why hasn't the patent holder at the very least provided Google with a DMCA notice? I guess because it's completely legal here, and there's a disclaimer on page: DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 patent holders for using their algorithm. It all depends on country and laws. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [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 -- 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 Documentation now on voip-info.org Wiki
+1 ! Great ! ___ -- 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] Software patents (was G723 on asterisk 1.4.1)
Atis Lezdins wrote: On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote: On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: It is completely illegal in any country that recognizes patents. You mean countries that recognize software patents, right? As resident of country where the file is hosted - yes we don't have software patents, they have been proposed to EU and reject few years ago. So by law - software is algorithm and can't be patented. In local laws we even are allowed to reverse-engineer software for needs of compatibility and interoperability. So, writing code for commercial codec and using it for interoperability with hardware devices (you purchased) is allowed by law. Damn, we even have a law that don't allow bittorrent trackers, as bittorrent file is considered breaking copyright law.. Ironic :p Please do NOT discuss ways to use unlicensed codecs on this list or any other forum provided by Digium. This has been discussed multiple times as to why not, and I don't feel like rehashing the argument again. I did not know you were a moderator on this list. contributory infringement What if I make a page that explains the patent issues and then provide a link to http://asterisk.hosting.lv/ from that site and only provide people on this list a link to my site? What if I provide a link to the Google search for asterisk g723? Where do we draw the line? If that site is so illegal, why hasn't it been taken down? Why hasn't the patent holder at the very least provided Google with a DMCA notice? I guess because it's completely legal here, and there's a disclaimer on page: DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 patent holders for using their algorithm. It all depends on country and laws. There are a few algorithmic speedup patents around, what can accelerate codecs like G.729 and G.723.1, and which are purely software patents. Most of the relevant patents are *not* software patents. Don't confuse software patent with something running on a computer. Patents applicable to speech coding are perfectly valid in the vast majority of countries. Certainly in all the EU countries. Regards, Steve ___ -- 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 Documentation now on voip-info.org Wiki
Josiah Bryan wrote: The script design supports plugin formatting as it stands. E.g. I can insert any formatting algorithm if anyone has any suggestions. Right now, the formatter script just does: #!/usr/bin/perl use strict; my $file = $ARGV[0]; print ~pp~\n; print `cat $file`; print ~/pp~\n; Any formatting can be added as desired - this was just a quick way to get the content online. Might I suggest including... print -=NOTE: These pages are automatically updated once per day/week/month/year/decade from the Asterisk subversion repository. Any changes made to this page will be automatically overwritten with the latest version from insert URL here.\n; ...at the beginning? May stop some nutters whining that you're continually overwriting their changes. ___ -- 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] How can Block a pri channel
Hi, why do not you simply delete them from zapata.conf and restart your PBX? Giorgio. Alireza Shokoienia wrote: Hi all, I'm new in astersik and like to know how can block a pri channel. It means I want block some channels on a pri link so nobody can occupy these channels, i.e channels 1 through 5 should be blocked. What is the Q931 message for blocking a channel? and can I see this message when I want use a protocol analyzer? thanks in advance Alireza ___ -- 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 -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ -- 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] Software patents (was G723 on asterisk 1.4.1)
On Wed, Oct 1, 2008 at 10:12 AM, Steve Underwood [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote: On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: It is completely illegal in any country that recognizes patents. You mean countries that recognize software patents, right? As resident of country where the file is hosted - yes we don't have software patents, they have been proposed to EU and reject few years ago. So by law - software is algorithm and can't be patented. In local laws we even are allowed to reverse-engineer software for needs of compatibility and interoperability. So, writing code for commercial codec and using it for interoperability with hardware devices (you purchased) is allowed by law. Damn, we even have a law that don't allow bittorrent trackers, as bittorrent file is considered breaking copyright law.. Ironic :p Please do NOT discuss ways to use unlicensed codecs on this list or any other forum provided by Digium. This has been discussed multiple times as to why not, and I don't feel like rehashing the argument again. I did not know you were a moderator on this list. contributory infringement What if I make a page that explains the patent issues and then provide a link to http://asterisk.hosting.lv/ from that site and only provide people on this list a link to my site? What if I provide a link to the Google search for asterisk g723? Where do we draw the line? If that site is so illegal, why hasn't it been taken down? Why hasn't the patent holder at the very least provided Google with a DMCA notice? I guess because it's completely legal here, and there's a disclaimer on page: DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 patent holders for using their algorithm. It all depends on country and laws. There are a few algorithmic speedup patents around, what can accelerate codecs like G.729 and G.723.1, and which are purely software patents. Most of the relevant patents are *not* software patents. Don't confuse software patent with something running on a computer. Patents applicable to speech coding are perfectly valid in the vast majority of countries. Certainly in all the EU countries. It seems that this have been discussed numerous times. http://lists.digium.com/pipermail/asterisk-users/2004-October/058136.html Does anybody have some more legal experence with this? Any courts? Negotiations? NDA? :p From what i've found, there's an EU directive regarding software patents, but it's full of legal terms. Maybe anyone can comment? http://www.europarl.europa.eu/commonpositions/2005/pdf/c6-0058-05_en.pdf Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [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 -- 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] GSM / 3g channel bank
More than 60% of our outbound calls are now to mobiles, so the time has come to whack in a gsm channel bank. Does anyone have any preference of bank ? Do you use a PRI or VOIP connection from the bank to asterisk ? Real-world experiences are so much better than marketing blurb ;) We currently have a TE412P with a free socket, so we have a choice either way. I am looking for up to 30 sims to be connected, and we are based in the UK. Any advice is gratefully received. Thanks Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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 Documentation now on voip-info.org Wiki
Rob Hillis wrote: Josiah Bryan wrote: Any formatting can be added as desired - this was just a quick way to get the content online. Might I suggest including... print -=NOTE: These pages are automatically updated once per day/week/month/year/decade from the Asterisk subversion repository. Any changes made to this page will be automatically overwritten with the latest version from insert URL here.\n; ...at the beginning? May stop some nutters whining that you're continually overwriting their changes. Good point - I'll get that in there after breakfast :-). Seriously, good point though. -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- 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] GSM / 3g channel bank
Portech has an excellent line of SIP-CDMA/GSM gateway products. We use their MV-378 for GSM and MV-372 for CDMA. We also have deployed them to quite a number of our customers with no issues. Jeff Johnson Director of Operations NeturallySpeaking, LLC (813) 774-3570 Direct (866) 448-0038 Toll Free (813) 569-2366 Fax sip://[EMAIL PROTECTED] http://www.neturallyspeaking.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Wednesday, October 01, 2008 7:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] GSM / 3g channel bank More than 60% of our outbound calls are now to mobiles, so the time has come to whack in a gsm channel bank. Does anyone have any preference of bank ? Do you use a PRI or VOIP connection from the bank to asterisk ? Real-world experiences are so much better than marketing blurb ;) We currently have a TE412P with a free socket, so we have a choice either way. I am looking for up to 30 sims to be connected, and we are based in the UK. Any advice is gratefully received. Thanks Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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: 8.0.173 / Virus Database: 270.7.5/1700 - Release Date: 9/30/2008 11:03 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] ATA for large networks
They and it make no sense in a conversation discussing half a dozen or more products.. On Mon, Sep 29, 2008 at 2:51 PM, C F [EMAIL PROTECTED] wrote: Oh, and not to mention, they are cheap. Very cheap in fact. On Mon, Sep 29, 2008 at 2:49 PM, C F [EMAIL PROTECTED] wrote: On Mon, Sep 29, 2008 at 11:15 AM, Steve Totaro [EMAIL PROTECTED] wrote: Huh, please try to form a complete thought I don't think I have to, since it's strictly an experience based questions, the answer I gave is what my experience has been. Grandstream can/will tell him that theirs works with some complete thought. And here is my thought, it just works, has all the features without a problem, and just don't need to be restarted. They will run and run and stay up unless you pull the power. Thanks, Steve Totaro On Mon, Sep 29, 2008 at 10:53 AM, C F [EMAIL PROTECTED] wrote: Channel Banks only On Mon, Sep 29, 2008 at 10:44 AM, Steve Totaro [EMAIL PROTECTED] wrote: Quintum Tenor AX. Just glance over the manual. The are far better and in my experience just as reliable as a channel bank. Thanks, Steve Totaro On Mon, Sep 29, 2008 at 10:26 AM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: The most reliable ATA is a channel bank. Vieri wrote: --- On Mon, 9/29/08, Sam Tam [EMAIL PROTECTED] wrote: Why not swap it all with just IP phone? That's because we have almost 400 analog phones already wired in our building (which is very large). So we need to take advantage of the wiring. Also, if we were to convert to an all-IP phone system (non-ATA), we would need to buy more ethernet switches (currently they're all full) and tunnel cables thtough ceilings and walls. In other words, it would cost a lot more than to simply buy ATAs. What I'm looking for however are STABLE, RELIABLE ATAs... -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. http://www.fnords.org/skillslist.html ___ -- 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 -- Thanks, Steve Totaro 1.888.777.1888 1.240.938.1212 (cell) ___ -- 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 -- Thanks, Steve Totaro 1.888.777.1888 1.240.938.1212 (cell) ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] GSM / 3g channel bank
On Wed, 1 Oct 2008, Jeff Johnson wrote: Portech has an excellent line of SIP-CDMA/GSM gateway products. We use their MV-378 for GSM and MV-372 for CDMA. We also have deployed them to quite a number of our customers with no issues. I've also used a Portech unit in the UK. (a 2-port GSM one) It's not perfect, but seems to work OK for outgoing calls which are LCR'd to the unit. (Easy in the UK as all mobile phone numbers start _07Z.) ISTR having to set some weird flags inside it to return CHANUNAVAIL rather then busy when both ports were in-use, so I could then try the default outbound channel when both ports were in-use. Gordon ___ -- 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] Call Center
I want to rekindle this conversation again, besides Asterisk Queues and VICIDIAL Call Center Suite. Are there any other options similar to Fonality. Has anyone intergerated SugarCRM? Thanks. On Thu, Jul 24, 2008 at 10:18 PM, Mohammad Salaque [EMAIL PROTECTED]wrote: live example : ;This is for XXX exten = 4455,1,Answer exten = 4455,n,Dial(SIP/GW3/XXX|75|mL(75)) exten = 4455,n,Hangup ; end of added ; lines added Techical T By for time base forwarding ;this is for S exten = 4456,1,Answer exten = 4456,n,Dial(SIP/GW3/X|75|mL(75)) exten = 4456,n,Hangup ; end of added exten = ,1,Answer exten = ,n,Wait(0) ;exten = ,n,Dial(SIP/RegS-1/02001|75|m) ;exten = ,n,Dial(SIP/80001|45|m) ;exten = ,n,Dial(SIP/GW2/998801671876162|75|mL(75)) exten = ,n,GoToIfTime(12:00-20:00|tue-sun|*|*?default|4455,1) exten = ,n,GoToIfTime(7:00-12:00|*|*|*?default|4456,1) exten = ,n,GoToIfTime(21:00-24:00|*|*|*?default|4456,1) exten = ,n,GoToIfTime(00:00-2:00|*|*|*?default|4456,1) exten = ,n,Dial(SIP/80001SIP/RegS-1/02001|45|m) ;exten = ,n,Goto(ext-local,80001,1) ;exten = ,n,Dial(SIP/Inter232/X|75|mL(60)) exten = ,n,Hangup Thanks On Tue, Jun 17, 2008 at 10:24 AM, broadband Voice [EMAIL PROTECTED] wrote: Is anyone using Asterisk as a call center. I want to be able to set it up for my office line, when calls come in after 7:00pm Est want a recording to says the office is closed and have about 5 phones that I want to use as an agent. Can anyone share their implementation? 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 -- M. Salaque VOIP Technician snphone.com voice : +12819712091 The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email. ___ -- 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 queue not play muscinhold or hangup
Any responses please. I'm interested in this as well. On Fri, Feb 8, 2008 at 2:51 AM, satish patel [EMAIL PROTECTED] wrote: Dear all I am going to setup Asterisk Call center solution and i have setup my queue and agent i have 2 SNOM ip phone but when i call to queue my agent phone is rining without musicnhold or when both phone is busy then i call to queue its directy hangup without musicnhole means my call not goes in to queue what is the problem my queue.conf [EMAIL PROTECTED] asterisk]# cat /etc/asterisk/queues.conf | grep -v ';' [general] persistentmembers = yes autofill = yes monitor-type = MixMonitor [] leavewhenempty = strict musiclass = default context=from-avaya strategy = rrmemory timeout = 20 retry = 1 wrapuptime=0 announce-frequency = 0 announce-holdtime = no persistentmembers = yes maxlen = 0 member = Agent/1001 member = Agent/1002 -- my agent.conf [general] persistentagents=yes [agents] ackcall=no musiconhold = default agent = 1001,1234,satish agent = 1002,1234,aman my dialplan ( extention.conf) exten = ,1,Answer exten = ,2,SetMusicOnHold(default) exten = ,3,Background(welcome) exten = ,4,Queue() PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org -- Looking for last minute shopping deals? Find them fast with Yahoo! Search.http://us.rd.yahoo.com/evt=51734/*http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- 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 -- 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] Debugging
First, I'm new to the list server format, so I'm trying to figure out the mechanics. I am installing an asterisk server for my SOHO. I'm working my way through Van Meggelen's book (just finished ch 4) and trying to get something up and working. Hardware is 1.8 GHz Pentium with 1 GB RAM OS is Fedora 9 I'm using Asterisk 1.6 I have a Linksys 941 SIP phone I will also be using ISDN for pstn and an engenius Wi-Fi SIP and a Grandstream 386 ATA, but that's future. Right now I just want asterisk to recognize the SIP after setting everything up as per ch 4 of the book. Everything seems to load fine. I don't see any obvious issues with either the SIP phone or asterisk, but the green light for line 1 doesn't come on as expected, and asterisk says its unmonitored. There are a couple of warnings for things like dundi that I don't think matter at this point. What sort of tests should I be running to narrow this down? The Linux firewall is off (my LAN has a firewall that protects me from the real world and all of this is inside it). SELinux is in permissive mode, as I'm not sure what I need to do to it so I got it out of the way. both the phone and the computer are on the same 10/100 switch. I don't think it gets much simpler. Thanks, Wilton ___ -- 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] New VICIDIAL astGUIclient Release: 2.0.4
Can I used Aastra phones as agents instead of web-base on astGUIclient-VICIDIAL suite: 2.0.4? Thanks. Our Asterisk is remote and call center will be using Aastra phones or Linksys ATA. On Mon, Dec 3, 2007 at 3:03 AM, Matt Florell [EMAIL PROTECTED] wrote: Hello, We've released another update to our astGUIclient-VICIDIAL suite: 2.0.4 http://astguiclient.sf.net/ The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the VICIDIAL call center suite and the astGUIclient client-side web app which extends your phone's functionality. This package is free and GPL. (the suite is not an asterisk configuration tool) This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks. For this release, we have focused on adding new features to inbound call handling such as custom music-on-hold, agent alert messages per inbound group and agent-rank call routing per skill as well as several other new administrative features. We have also tested the suite on Asterisk versions through 1.2.24. All client web-apps and administration pages are available in English, Spanish, Greek and German, with rough translations of French, Polish, Italian, Portuguese and Brazillian Portuguese for the client web-apps only. Check out the project blog for more information: http://astguiclient.blogspot.com Let me know what you think. 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 ___ -- 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] RTP sent before the INVITE ACK (for voicemail app)
Hello, With asterisk 1.4.11, I am calling AGI exec voicemail upon a SIP INVITE invite - asterisk - 100 - 200 - RTP ACK - ... asterisk is sending the RTP for the greeting before the original invite is ACK-ed (confirmed with a tcpdump) as if playing the prompt as soon as it is received from the AGI. I don't see any 183 so I don't think early media should apply. CLI output does not show any error that I see. Is there any reason other than a SIP 183 that would trigger this and isn't asterisk supposed to ACK/answer the channel before playing any prompt? Thanks, Sebastien. ___ -- 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 in VM.
Does anyone have any perspective on how well Asterisk performs and scales inside a Xen hypervisor environment? I tried on many different pieces of hardware with various recent Xen versions and it always had some level of unpredictability and was not as reliable as running on bare hardware. I wouldn't do it for production but it was fine for testing (sort of :). This was of course w/ ztdummy in a pure sip env. jlc ___ -- 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] Software patents (was G723 on asterisk 1.4.1)
Atis Lezdins wrote: On Wed, Oct 1, 2008 at 10:12 AM, Steve Underwood [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote: On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: It is completely illegal in any country that recognizes patents. You mean countries that recognize software patents, right? As resident of country where the file is hosted - yes we don't have software patents, they have been proposed to EU and reject few years ago. So by law - software is algorithm and can't be patented. In local laws we even are allowed to reverse-engineer software for needs of compatibility and interoperability. So, writing code for commercial codec and using it for interoperability with hardware devices (you purchased) is allowed by law. Damn, we even have a law that don't allow bittorrent trackers, as bittorrent file is considered breaking copyright law.. Ironic :p Please do NOT discuss ways to use unlicensed codecs on this list or any other forum provided by Digium. This has been discussed multiple times as to why not, and I don't feel like rehashing the argument again. I did not know you were a moderator on this list. contributory infringement What if I make a page that explains the patent issues and then provide a link to http://asterisk.hosting.lv/ from that site and only provide people on this list a link to my site? What if I provide a link to the Google search for asterisk g723? Where do we draw the line? If that site is so illegal, why hasn't it been taken down? Why hasn't the patent holder at the very least provided Google with a DMCA notice? I guess because it's completely legal here, and there's a disclaimer on page: DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 patent holders for using their algorithm. It all depends on country and laws. There are a few algorithmic speedup patents around, what can accelerate codecs like G.729 and G.723.1, and which are purely software patents. Most of the relevant patents are *not* software patents. Don't confuse software patent with something running on a computer. Patents applicable to speech coding are perfectly valid in the vast majority of countries. Certainly in all the EU countries. It seems that this have been discussed numerous times. http://lists.digium.com/pipermail/asterisk-users/2004-October/058136.html Does anybody have some more legal experence with this? Any courts? Negotiations? NDA? :p From what i've found, there's an EU directive regarding software patents, but it's full of legal terms. Maybe anyone can comment? http://www.europarl.europa.eu/commonpositions/2005/pdf/c6-0058-05_en.pdf You're back to talking about software patents again. People love to do that, in the same spirit that schoolboys cross their fingers in the hope it absolves them from something. Would you care to look through the patents which the G.729 patent pool licences, and try to find any software patents amongst them? Regards, Steve ___ -- 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] GSM / 3g channel bank
On Wed, Oct 01, 2008 at 12:53:41PM +0100, Julian Lyndon-Smith wrote: More than 60% of our outbound calls are now to mobiles, so the time has come to whack in a gsm channel bank. Does anyone have any preference of bank ? Do you use a PRI or VOIP connection from the bank to asterisk ? Real-world experiences are so much better than marketing blurb ;) We currently have a TE412P with a free socket, so we have a choice either way. I am looking for up to 30 sims to be connected, and we are based in the UK. Any advice is gratefully received. Are you providing any kind of service to 3rd parties? If so you are NOT allowed to run a GSM gateway. If it's purely your own traffic i.e. say company PBX to mobile traffic then it is allowed. Ofcom ruled on this a while back. Saying that I've used a Portech GSM gateway on SIP and it works well. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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
Re: [asterisk-users] GSM / 3g channel bank
Hi Steve, Steve Kennedy wrote: On Wed, Oct 01, 2008 at 12:53:41PM +0100, Julian Lyndon-Smith wrote: More than 60% of our outbound calls are now to mobiles, so the time has come to whack in a gsm channel bank. Does anyone have any preference of bank ? Do you use a PRI or VOIP connection from the bank to asterisk ? Real-world experiences are so much better than marketing blurb ;) We currently have a TE412P with a free socket, so we have a choice either way. I am looking for up to 30 sims to be connected, and we are based in the UK. Any advice is gratefully received. Are you providing any kind of service to 3rd parties? If so you are NOT allowed to run a GSM gateway. No, it's purely for our own call center. If it's purely your own traffic i.e. say company PBX to mobile traffic then it is allowed. Ofcom ruled on this a while back. Yeah. They also ruled that last week a call that never rings on the customer side before it is hung up is considered a silent call ... Saying that I've used a Portech GSM gateway on SIP and it works well. Thanks for the info. Steve __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] How can Block a pri channel
Giorgio Incantalupo wrote: Hi, why do not you simply delete them from zapata.conf and restart your PBX? Because that simply doesn't acheive what he's wanting to achieve. On PRI circuits you can dynamically enable and disable circuits at the data-link level. Whether this can be achieved with Asterisk or not, I don't know. ___ -- 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] Sip Header Help
Dear List: I need to make a sip phone (spa942) answer a call but the phone must no ring. The user only has to show the callerId on the phone screen without any sound. How could I make that in asterisk? I tried to use Sip headers but I do not know how must I say the phone don't ring when received, only shows the callerID of the call. How could I do that with sip header? Which sip header should I send the phone to change the callerID of the call? Do you know any other way to ghet that. Any clue will be wellcomed. Thanks for your answer. 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
Re: [asterisk-users] GSM / 3g channel bank
On Wed, 1 Oct 2008, Steve Kennedy wrote: On Wed, Oct 01, 2008 at 12:53:41PM +0100, Julian Lyndon-Smith wrote: More than 60% of our outbound calls are now to mobiles, so the time has come to whack in a gsm channel bank. Does anyone have any preference of bank ? Do you use a PRI or VOIP connection from the bank to asterisk ? Real-world experiences are so much better than marketing blurb ;) We currently have a TE412P with a free socket, so we have a choice either way. I am looking for up to 30 sims to be connected, and we are based in the UK. Any advice is gratefully received. Are you providing any kind of service to 3rd parties? If so you are NOT allowed to run a GSM gateway. If it's purely your own traffic i.e. say company PBX to mobile traffic then it is allowed. Ofcom ruled on this a while back. Saying that I've used a Portech GSM gateway on SIP and it works well. Does anyone know if this is true in the States? I run a prepaid card service, and never thought about trying to use SIM cards for termination :) j ___ -- 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] RTP sent before the INVITE ACK (for voicemail app)
CLI output does not show any error that I see. Is there any reason other than a SIP 183 that would trigger this and isn't asterisk supposed to ACK/answer the channel before playing any prompt? Asterisk wil start the audio as soon as it sends back the 200 Ok response it doesn't wait for the ACK. Most SIP servers will work like that. The matching of ACK requests to a SIP transaction is not a particulalrly robust mechanism (for instance if a user agent puts its IP address in the Call-ID and a SIP ALG fiddles with the SIP packet for INVITEs but ignores ACKs then there will be a mismatch. This happens more frequently then you would think) so sending RTP after an OK response is the correct thing to do. I think Asterisk will actually cut off the call after 32s if it doesn't get an ACK which is not such a great idea but that may have been changed in later versions. The arrival of an RTP packet from the remote end should be used as the definitive indication of an answered call not the ACK. Regards, Greyman. ___ -- 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] Ongoing calls with SIPPEER, curcalls
Hello,
From http://www.voip-info.org/wiki/view/Asterisk+func+sippeer I understood I
could use SIPPEER and curcalls parameter to get the number of ongoing calls
passed of received by a given peer.
Strangely, in my system, returned value remains equal to 0, even when the
targeted peer is oncall with another one.
exten = _753X,n,Set(foo=${SIPPEER(${EXTEN}:curcalls)})
exten = _753X,n,Set(foo2=${SIPPEER(${EXTEN}:limit)})
exten = _753X,n,NoOp(${foo} ${foo2})
Replies are 0 and 5.
Am I missing something ?
Cheers
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Re: [asterisk-users] Ongoing calls with SIPPEER, curcalls
Olivier wrote: Replies are 0 and 5. Am I missing something ? show function SIPPEER curcalls Current amount of calls __Only available if call-limit is set__ Have you set call-limit? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] SIP users limit
People, what is the limit of SIP users in Asterisk 1.4.x ??? What the main difference if I use OpenSER ??? Thank you. A.F. ___ -- 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 in VM.
My experience is very limited, but you asked for any perspective, so... I put an Asterisk with freePBX on a linode server (linode.com), just to play with it a few months ago. I can say that it worked to the point of being able to dial out with my Polycom phone on a FiOS connection, through the * box, and a SIP termination service like Vitelity, and to receive calls in the other direction. No problems with that, and kinda cool to be able to throw a virtual PBX out there with so little expense. I did not stress test it, nor did I examine resource usage to gain perspective on scalability. More of a proof of concept. One issue that comes up with regard to this is about timing sources for MOH, etc. Related to this and of general use to know, I believe one can associate PCI cards with particular VMs, but it's been a few months since I configured a Xen box of my own, so the details have already fled from my feeble brain... But I hope that's helpful. Alex Balashov wrote: Does anyone have any perspective on how well Asterisk performs and scales inside a Xen hypervisor environment? Obviously, the answer depends largely on what sort of hardware it's running on, whether it's in PAE mode, whether it's a newer CPU that has some paravirtualisation instruction sets available to assist it, how much memory is allocated to each VM, and other architectural considerations. Any perspective would be helpful, however. smime.p7s Description: S/MIME Cryptographic Signature ___ -- 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] G723 on asterisk 1.4.1
On Tuesday 30 September 2008 22:34:30 Andrew Joakimsen wrote: On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: It is completely illegal in any country that recognizes patents. You mean countries that recognize software patents, right? Correct. Please do NOT discuss ways to use unlicensed codecs on this list or any other forum provided by Digium. This has been discussed multiple times as to why not, and I don't feel like rehashing the argument again. I did not know you were a moderator on this list. Currently, no, but I can become a moderator if that's what the problem warrants. contributory infringement What if I make a page that explains the patent issues and then provide a link to [[[LINK REMOVED]]] from that site and only provide people on this list a link to my site? What if I provide a link to the Google search for asterisk g723? Where do we draw the line? If that site is so illegal, why hasn't it been taken down? Why hasn't the patent holder at the very least provided Google with a DMCA notice? The site is hosted internationally, in a country that does not recognize software patents, and Google's right to index the web is not in dispute. -- Tilghman ___ -- 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] Software patents (was G723 on asterisk 1.4.1)
On Wed, Oct 1, 2008 at 5:09 PM, Steve Underwood [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Wed, Oct 1, 2008 at 10:12 AM, Steve Underwood [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote: On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: It is completely illegal in any country that recognizes patents. You mean countries that recognize software patents, right? As resident of country where the file is hosted - yes we don't have software patents, they have been proposed to EU and reject few years ago. So by law - software is algorithm and can't be patented. In local laws we even are allowed to reverse-engineer software for needs of compatibility and interoperability. So, writing code for commercial codec and using it for interoperability with hardware devices (you purchased) is allowed by law. Damn, we even have a law that don't allow bittorrent trackers, as bittorrent file is considered breaking copyright law.. Ironic :p Please do NOT discuss ways to use unlicensed codecs on this list or any other forum provided by Digium. This has been discussed multiple times as to why not, and I don't feel like rehashing the argument again. I did not know you were a moderator on this list. contributory infringement What if I make a page that explains the patent issues and then provide a link to http://asterisk.hosting.lv/ from that site and only provide people on this list a link to my site? What if I provide a link to the Google search for asterisk g723? Where do we draw the line? If that site is so illegal, why hasn't it been taken down? Why hasn't the patent holder at the very least provided Google with a DMCA notice? I guess because it's completely legal here, and there's a disclaimer on page: DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 patent holders for using their algorithm. It all depends on country and laws. There are a few algorithmic speedup patents around, what can accelerate codecs like G.729 and G.723.1, and which are purely software patents. Most of the relevant patents are *not* software patents. Don't confuse software patent with something running on a computer. Patents applicable to speech coding are perfectly valid in the vast majority of countries. Certainly in all the EU countries. It seems that this have been discussed numerous times. http://lists.digium.com/pipermail/asterisk-users/2004-October/058136.html Does anybody have some more legal experence with this? Any courts? Negotiations? NDA? :p From what i've found, there's an EU directive regarding software patents, but it's full of legal terms. Maybe anyone can comment? http://www.europarl.europa.eu/commonpositions/2005/pdf/c6-0058-05_en.pdf You're back to talking about software patents again. People love to do that, in the same spirit that schoolboys cross their fingers in the hope it absolves them from something. Would you care to look through the patents which the G.729 patent pool licences, and try to find any software patents amongst them? Because it's one directive regulating software and mathematical/algorithmic patents. Personally, I don't use G.729 at all, i'm just curios about this. If you would point me, i would gladly take a look at this patent list, for now my searches were unsuccessful. I was also asking somebody for some legal experience with this, as theory and practical application of patent laws may differ :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [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 -- 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] [Fwd: asterisk-users Digest, Vol 51, Issue 2]
From: Joseph L. Casale [EMAIL PROTECTED] Does anyone have any perspective on how well Asterisk performs and scales inside a Xen hypervisor environment? I tried on many different pieces of hardware with various recent Xen versions and it always had some level of unpredictability and was not as reliable as running on bare hardware. I wouldn't do it for production but it was fine for testing (sort of :). All other things being equal, certainly the bare HW will win out. I'd like to also note that Xen provides a mechanism to dedicate a CPU core to a virtual machine in a system appropriately equipped. Might be useful. As to whether all critical sources of contention can be controlled adequately to achieve an equivalent or sufficiently robust environment for Asterisk -- I can't say authoritatively. It's reasonable to think it might be possible. smime.p7s Description: S/MIME Cryptographic Signature ___ -- 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] New VICIDIAL astGUIclient Release: 2.0.4
Hello, I have never tried using Aastra phones as user agent. If they support Javascript and AJAX then it should work. VICIDIAL is tested with IE, Firefox, Opera and Safari. At the very least they may be able to use the remote agent interface that does not use Javascript, but there is reduced functionality as compared to the full agent interface. Thanks, MATT--- On 10/1/08, broadband Voice [EMAIL PROTECTED] wrote: Can I used Aastra phones as agents instead of web-base on astGUIclient-VICIDIAL suite: 2.0.4? Thanks. Our Asterisk is remote and call center will be using Aastra phones or Linksys ATA. On Mon, Dec 3, 2007 at 3:03 AM, Matt Florell [EMAIL PROTECTED] wrote: Hello, We've released another update to our astGUIclient-VICIDIAL suite: 2.0.4 http://astguiclient.sf.net/ The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the VICIDIAL call center suite and the astGUIclient client-side web app which extends your phone's functionality. This package is free and GPL. (the suite is not an asterisk configuration tool) This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks. For this release, we have focused on adding new features to inbound call handling such as custom music-on-hold, agent alert messages per inbound group and agent-rank call routing per skill as well as several other new administrative features. We have also tested the suite on Asterisk versions through 1.2.24. All client web-apps and administration pages are available in English, Spanish, Greek and German, with rough translations of French, Polish, Italian, Portuguese and Brazillian Portuguese for the client web-apps only. Check out the project blog for more information: http://astguiclient.blogspot.com Let me know what you think. 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 ___ -- 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 in VM.
Alex Balashov wrote: Does anyone have any perspective on how well Asterisk performs and scales inside a Xen hypervisor environment I did asterisk in xen recently (ubuntu hardy, xen 3.2). 10 sip users, 1 fxs, 1 e1. used xen pci passthrough for digium cards. no problems at this day. same machine hosts vms for fileserver, sms-gateway, imap, smtp, squid and router Alexander ___ -- 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] Ongoing calls with SIPPEER, curcalls
2008/10/1 Doug Lytle [EMAIL PROTECTED]
Olivier wrote:
Replies are 0 and 5.
Am I missing something ?
show function SIPPEER
curcalls Current amount of calls
__Only available if call-limit is set__
Have you set call-limit?
yes : that's why I displayed both values here :
exten = _753X,n,Set(foo=${SIPPEER(${EXTEN}:curcalls)})
exten = _753X,n,Set(foo2=${SIPPEER(${EXTEN}:limit)})
exten = _753X,n,NoOp(${foo} ${foo2})
call-limit is valued to 5 while curcalls remains equal to 0
or maybe I'm missing something obvious ...
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] RTP sent before the INVITE ACK (for voicemail app)
Thanks, in my case though it looks like the originating party (polycom softphone) is hearing a clipped voicemail prompt because of that; should I look into having that fixed into their firmware? As a workaround, I was thinking to just add a few seconds delay in app_voicemail, or wait through AGI before calling voicemail, makes sense? Date: Wed, 1 Oct 2008 15:43:37 +0100 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] RTP sent before the INVITE ACK (for voicemail app) CLI output does not show any error that I see. Is there any reason other than a SIP 183 that would trigger this and isn't asterisk supposed to ACK/answer the channel before playing any prompt? Asterisk wil start the audio as soon as it sends back the 200 Ok response it doesn't wait for the ACK. Most SIP servers will work like that. The matching of ACK requests to a SIP transaction is not a particulalrly robust mechanism (for instance if a user agent puts its IP address in the Call-ID and a SIP ALG fiddles with the SIP packet for INVITEs but ignores ACKs then there will be a mismatch. This happens more frequently then you would think) so sending RTP after an OK response is the correct thing to do. I think Asterisk will actually cut off the call after 32s if it doesn't get an ACK which is not such a great idea but that may have been changed in later versions. The arrival of an RTP packet from the remote end should be used as the definitive indication of an answered call not the ACK. Regards, Greyman. ___ -- 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] Ongoing calls with SIPPEER, curcalls
Olivier wrote:
yes : that's why I displayed both values here :
exten = _753X,n,Set(foo=${SIPPEER(${EXTEN}:curcalls)})
exten = _753X,n,Set(foo2=${SIPPEER(${EXTEN}:limit)})
exten = _753X,n,NoOp(${foo} ${foo2})
Look into your sip.conf and search on limit.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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[asterisk-users] ietf-sipping-config-framework
Hi, Is there any plan to support this http://tools.ietf.org/html/draft-ietf-sipping-config-framework-15 within or outside Asterisk, using AMI for instance ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ongoing calls with SIPPEER, curcalls
2008/10/1 Doug Lytle [EMAIL PROTECTED]
Olivier wrote:
yes : that's why I displayed both values here :
exten = _753X,n,Set(foo=${SIPPEER(${EXTEN}:curcalls)})
exten = _753X,n,Set(foo2=${SIPPEER(${EXTEN}:limit)})
exten = _753X,n,NoOp(${foo} ${foo2})
Look into your sip.conf and search on limit.
I can see in sip.conf :
call-limit=5
So, when Set(foo2=${SIPPEER(${EXTEN}:limit)}) replies 5, it replies to
correct value.
The strange thing is with curcalls ...
As I'm using the same method to get its value, either :
- curcalls is not set to what I was thinking (I misunderstood its definition
in voip-info.org, as I comply to call-limit setting requirement)
- something else
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] Ongoing calls with SIPPEER, curcalls
Olivier wrote: - curcalls is not set to what I was thinking (I misunderstood its definition in voip-info.org http://voip-info.org, as I comply to call-limit setting requirement) - something else My very basic testing, I'm not able to get a value either. I won't have access to my testing system until later on this evening. I'll give it another try then. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] zap destroy
- Jeff LaCoursiere [EMAIL PROTECTED] wrote: One of my clients today had a POTS line with a bad punch, and no dialtone. I used zap destroy channel x remotely to keep it from being used to send outbound calls, which worked fine. Line repunched, ready again to use, but how do I undestroy the channel? In the end I kicked everyone off with zap restart (which for some reason I had to do twice). Is there are a more elegant method to deal with this kind of issue? Cheers, j Nope, that's the best you can do without restarting Asterisk. Is requiring two restarts reproducible? I'd really like to see console output with verbosity and debug set to 4 on chan_dahdi, preferably while only using zap channels. Jeff ___ -- 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] GSM / 3g channel bank
If you are after something budget type we got a 32 SIM with only 8 active sims at anyone time so you can use each sim in term and effectively use up all the free miuntes, If you are interested please visit cyber-telecom.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Wednesday, October 01, 2008 7:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] GSM / 3g channel bank More than 60% of our outbound calls are now to mobiles, so the time has come to whack in a gsm channel bank. Does anyone have any preference of bank ? Do you use a PRI or VOIP connection from the bank to asterisk ? Real-world experiences are so much better than marketing blurb ;) We currently have a TE412P with a free socket, so we have a choice either way. I am looking for up to 30 sims to be connected, and we are based in the UK. Any advice is gratefully received. Thanks Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] How can Block a pri channel
Rob Hillis wrote: Giorgio Incantalupo wrote: Hi, why do not you simply delete them from zapata.conf and restart your PBX? Because that simply doesn't acheive what he's wanting to achieve. On PRI circuits you can dynamically enable and disable circuits at the data-link level. Whether this can be achieved with Asterisk or not, I don't know. This may be what you are looking for: http://bugs.digium.com/view.php?id=3450 -- Sean Bright [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
Re: [asterisk-users] How can Block a pri channel
- Sean Bright [EMAIL PROTECTED] wrote: This may be what you are looking for: http://bugs.digium.com/view.php?id=3450 Sean is correct, the branches specified in issue 3450 provide this functionality. There are currently issues with NFAS configurations. If you use the branches specified in the last comments of issue 3450, please provide some feedback on http://bugs.digium.com/view.php?id=3450 thanks, -Dwayne ___ -- 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 - Failover System
Dear Group, I would like to know the best configuration to do a system with failover (Asterisk- T1's) Users: 120 Channels: 2T1's Thanks in advance for your help, Nelson ___ -- 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 - Failover System
Depends on your conditions for failure. If you are simply looking to get around hardware failures or other physical conditions which result in the T1s going down, you can use a reliable piece of ISDN gateway equipment in front of the Asterisk boxes that pushes SIP out the other end and configure it to fail over calls to a second VoIP peer. It creates a single point of failure, but one which is far, far more unlikely to fail than PC hardware. Or, you can use a DS1-level DACS that can protection switch the T1s to another port. Ultimately, your failover options are limited if you have T1s physically going into an Asterisk box with a T1 card. Your best bet is to somehow separate the two components; then, you can get the proper redundancy measures implemented on both sides in ways that are most native and technologically appropriate, instead of coupling them in a way that makes them difficult to divest. Nelson Granados wrote: Dear Group, I would like to know the best configuration to do a system with failover (Asterisk- T1’s) Users: 120 Channels: 2T1’s Thanks in advance for your help, Nelson ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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 - Failover System
For asterisk you can use heartbeat. regarding T1, you will need some thing out outside Asterisk server. Any reason you want to go for T1, not true VoIP? Jai http://www.didforsale.com/ *Buy SIP DIDs all Over US at low cost, unlimited minutes http://www.didforsale.com; On Wed, Oct 1, 2008 at 2:24 PM, Nelson Granados [EMAIL PROTECTED]wrote: Dear Group, I would like to know the best configuration to do a system with failover (Asterisk- T1's) Users: 120 Channels: 2T1's Thanks in advance for your help, Nelson ___ -- 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 in VM.
On Tue, 2008-09-30 at 19:05 -0400, Alex Balashov wrote: Does anyone have any perspective on how well Asterisk performs and scales inside a Xen hypervisor environment? Obviously, the answer depends largely on what sort of hardware it's running on, whether it's in PAE mode, whether it's a newer CPU that has some paravirtualisation instruction sets available to assist it, how much memory is allocated to each VM, and other architectural considerations. Any perspective would be helpful, however. We have been doing some test. To avoid single point of failure (as musch as possible) we had mysql, ldap and asterisk14 and asterisk16 in separate virtual machines. No problem what so ever. DOM-u's are easy to scale up, considering mem or cpu. Would suggest to have either channel-banks/ata's or PRI-boards in a separate machine(s). never got the pci forwarding from a hypervisor to a dom-U properly working Hans ___ -- 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 in VM.
Its very iffy to say the least. Jeff Johnson Director of Operations sip://[EMAIL PROTECTED] http://www.neturallyspeaking.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hans Witvliet Sent: Wednesday, October 01, 2008 6:07 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk in VM. On Tue, 2008-09-30 at 19:05 -0400, Alex Balashov wrote: Does anyone have any perspective on how well Asterisk performs and scales inside a Xen hypervisor environment? Obviously, the answer depends largely on what sort of hardware it's running on, whether it's in PAE mode, whether it's a newer CPU that has some paravirtualisation instruction sets available to assist it, how much memory is allocated to each VM, and other architectural considerations. Any perspective would be helpful, however. We have been doing some test. To avoid single point of failure (as musch as possible) we had mysql, ldap and asterisk14 and asterisk16 in separate virtual machines. No problem what so ever. DOM-u's are easy to scale up, considering mem or cpu. Would suggest to have either channel-banks/ata's or PRI-boards in a separate machine(s). never got the pci forwarding from a hypervisor to a dom-U properly working Hans ___ -- 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: 8.0.173 / Virus Database: 270.7.5/1702 - Release Date: 10/1/2008 9:05 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] Software patents (was G723 on asterisk 1.4.1)
If you would point me, i would gladly take a look at this patent list, for now my searches were unsuccessful. The ITU maintains a list of IPR (Intellectual Property Rights) claims for various technologies. Check it out: http://www.itu.int/ipr/IPRSearch.aspx?iprtype=PS On the left-hand side there's a search box, plus you can select G.729 (or one of the many derivatives thereof) from the recommendations drop-down list. When I select G.729 and click Search I get back a list of 52 items, most of which seem to be patents that have at least one claim related to this codec. (I see lots of references to stuff like CS-ACELP and other super-geekish acronyms that only smart people like Steve Underwood actually understand!:) IANAL but it looks like a lot of people have their hands out expecting payment for people using G.729: www.sipro.com, e.g. Happy researching, MC ___ -- 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 - Failover System
You can use two OpenSer boxen with heartbeat and the dispatch module for load balancing if you need it, and failover, in front of a couple of Asterisk boxen connected to a Redfone device (TDMoE). Thanks, Steve Totaro On Wed, Oct 1, 2008 at 5:40 PM, Jai Rangi [EMAIL PROTECTED] wrote: For asterisk you can use heartbeat. regarding T1, you will need some thing out outside Asterisk server. Any reason you want to go for T1, not true VoIP? Jai http://www.didforsale.com/ *Buy SIP DIDs all Over US at low cost, unlimited minutes http://www.didforsale.com; On Wed, Oct 1, 2008 at 2:24 PM, Nelson Granados [EMAIL PROTECTED] wrote: Dear Group, I would like to know the best configuration to do a system with failover (Asterisk- T1's) Users: 120 Channels: 2T1's Thanks in advance for your help, Nelson ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Software patents (was G723 on asterisk 1.4.1)
I own this combination of 1s and 0s. 111010010010101001001. LOL On Wed, Oct 1, 2008 at 6:50 PM, Michael Collins [EMAIL PROTECTED]wrote: If you would point me, i would gladly take a look at this patent list, for now my searches were unsuccessful. The ITU maintains a list of IPR (Intellectual Property Rights) claims for various technologies. Check it out: http://www.itu.int/ipr/IPRSearch.aspx?iprtype=PS On the left-hand side there's a search box, plus you can select G.729 (or one of the many derivatives thereof) from the recommendations drop-down list. When I select G.729 and click Search I get back a list of 52 items, most of which seem to be patents that have at least one claim related to this codec. (I see lots of references to stuff like CS-ACELP and other super-geekish acronyms that only smart people like Steve Underwood actually understand!:) IANAL but it looks like a lot of people have their hands out expecting payment for people using G.729: www.sipro.com, e.g. Happy researching, MC ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 - Failover System
Yep. OpenSER often has an instrumental role to play here. Steve Totaro wrote: You can use two OpenSer boxen with heartbeat and the dispatch module for load balancing if you need it, and failover, in front of a couple of Asterisk boxen connected to a Redfone device (TDMoE). Thanks, Steve Totaro On Wed, Oct 1, 2008 at 5:40 PM, Jai Rangi [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: For asterisk you can use heartbeat. regarding T1, you will need some thing out outside Asterisk server. Any reason you want to go for T1, not true VoIP? Jai http://www.didforsale.com/ *Buy SIP DIDs all Over US at low cost, unlimited minutes http://www.didforsale.com http://www.didforsale.com/ On Wed, Oct 1, 2008 at 2:24 PM, Nelson Granados [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Dear Group, I would like to know the best configuration to do a system with failover (Asterisk- T1's) Users: 120 Channels: 2T1's Thanks in advance for your help, Nelson ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] zap destroy
On Oct 1, 2008, at 11:39 AM, Jeff Peeler wrote: Nope, that's the best you can do without restarting Asterisk. Is requiring two restarts reproducible? I'd really like to see console output with verbosity and debug set to 4 on chan_dahdi, preferably while only using zap channels. For me, yes. Every single time I do a zap restart I have to do it twice. If I execute them REALLY fast I have to do it 3 times. I am using 1.4.20.1 with chan_zap still, but I will try to produce you a copy of the log this weekend (doing some phone maintenance anyway). I have experienced this for as long as I can remember, and I know bad form, but just have never gotten around to filing a bug report on it. One of those things you never remember until you do 'zap restart' and it fails and then you do it again and go whew, that was close. Daniel Jeff ___ -- 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] Software patents (was G723 on asterisk 1.4.1)
Michael Collins wrote: IANAL but it looks like a lot of people have their hands out expecting payment for people using G.729: www.sipro.com, e.g. Heh. :) Well, I have my hands out expecting a Treasury bailout... -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Software patents (was G723 on asterisk 1.4.1)
IANAL but it looks like a lot of people have their hands out expecting payment for people using G.729: www.sipro.com, e.g. Happy researching, MC I keep a stash of 1,000 500mg sipro. Gotta be prepared these days -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Software patents (was G723 on asterisk 1.4.1)
I wonder if they've got patents on various strains of Anthrax... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, October 01, 2008 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Software patents (was G723 on asterisk 1.4.1) IANAL but it looks like a lot of people have their hands out expecting payment for people using G.729: www.sipro.com, e.g. Happy researching, MC I keep a stash of 1,000 500mg sipro. Gotta be prepared these days -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] zap destroy
On Wed, 1 Oct 2008, Daniel Hazelbaker wrote: On Oct 1, 2008, at 11:39 AM, Jeff Peeler wrote: Nope, that's the best you can do without restarting Asterisk. Is requiring two restarts reproducible? I'd really like to see console output with verbosity and debug set to 4 on chan_dahdi, preferably while only using zap channels. For me, yes. Every single time I do a zap restart I have to do it twice. If I execute them REALLY fast I have to do it 3 times. I am using 1.4.20.1 with chan_zap still, but I will try to produce you a copy of the log this weekend (doing some phone maintenance anyway). I have experienced this for as long as I can remember, and I know bad form, but just have never gotten around to filing a bug report on it. One of those things you never remember until you do 'zap restart' and it fails and then you do it again and go whew, that was close. It has happened to me twice - the only two times I have ever used it. Is there a better way to busy out a port so it won't be used? Cheers, j ___ -- 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 - Failover System
Yes Redfone will do the T1 failover. Openser? for 120 user? I would not do that. This would be an extra layer to configure, support, maintain and one more layer to debug if things go wrong. Its like spending a Dollar when you can be done with a quarter. (my 2 cents) Jai *Buy SIP DIDs all Over US at low cost, unlimited minutes http://www.didforsale.com; On Wed, Oct 1, 2008 at 4:00 PM, Steve Totaro [EMAIL PROTECTED] wrote: You can use two OpenSer boxen with heartbeat and the dispatch module for load balancing if you need it, and failover, in front of a couple of Asterisk boxen connected to a Redfone device (TDMoE). Thanks, Steve Totaro On Wed, Oct 1, 2008 at 5:40 PM, Jai Rangi [EMAIL PROTECTED] wrote: For asterisk you can use heartbeat. regarding T1, you will need some thing out outside Asterisk server. Any reason you want to go for T1, not true VoIP? Jai http://www.didforsale.com/ *Buy SIP DIDs all Over US at low cost, unlimited minutes http://www.didforsale.com; On Wed, Oct 1, 2008 at 2:24 PM, Nelson Granados [EMAIL PROTECTED] wrote: Dear Group, I would like to know the best configuration to do a system with failover (Asterisk- T1's) Users: 120 Channels: 2T1's Thanks in advance for your help, Nelson ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] no audio, firewall problem?
Hello, I am runing asterisk on a embedded linux and am having some RTP audio issues at the beginning of the call: the comfort noise packet seems to be opening the pinhole in the firewall though I don't understand why it is not already opened. Then audio is then transferred correctly between caller and callee through the asterisk bridge. The SIP INVITE is received on a WAN interface and then I dial out to another SIP channel through the same interface. CLI output with RTP debug shows that Packet2Packet is only started and RTP is only sent by asterisk after the first rtpkeepalive timeout. If I sniff at a mirroring port in the network I can see the first RTP packet going from my caller to the asterisk server yet it seems that it is never received (or it never reaches) asterisk (it is a direct route). All firewall rules on the asterisk box are setup for the range of ports defined by rtp.conf (10k-11k in mycase); that is consistent with the SDP signaling generated by asterisk for the INVITE OUT and for the 200 OK back to the caller in the media description attribute. Watching iptables live activation does not show any RTP packet blocked at the beginning of the call. netstat shows: netstat -an | grep udp | grep 10 netstat: no support for 'AF INET6 (tcp)' on this system netstat: no support for 'AF INET6 (udp)' on this system netstat: no support for 'AF INET6 (raw)' on this system udp0 0 216.54.141.148:105540.0.0.0:* udp0 0 216.54.141.148:105550.0.0.0:* udp0 0 216.54.141.148:101020.0.0.0:* udp0 0 216.54.141.148:101030.0.0.0:* as I am using bindaddr=0.0.0.0 in the sip.conf. I have multiple NICs on that box, could it be a problem or ...? Thanks for any suggestion, Sebastien. ___ -- 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] rebooting snoms in 1.6
With Asterisk 1.4 I could use commands like: /usr/sbin/asterisk -rx sip notify reboot-snom mjc_home to reboot a snom phone. Now, with 1.6, when I try that, I get: Unable to find notify type 'reboot-snom' Command 'sip notify reboot-snom mjc_home' failed. Do I need to add some magic to sip_notify.conf? I haven't quite figured out how to make it work. - Mike ___ -- 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 - Failover System
On Wed, 1 Oct 2008, Jai Rangi wrote: Openser? for 120 user? I would not do that. This would be an extra layer to configure, support, maintain and one more layer to debug if things go wrong. Its like spending a Dollar when you can be done with a quarter. (my 2 cents) Jai *Buy *** * I've used OpenSER to front-end Asterisk several times and am always pleased with the result. The configuration is trivial for my needs and I find it to be rock solid. I love that you can restart it without interrupting calls in flight. I've never had to use it in a fail-over situation -- yet :) From a system administration standpoint, it is a big win. I can take a system out of production in a truly graceful fashion for updating the software on my Asterisk boxes. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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
Google with site:lists.digium.com which will return answers only from archive. On Mon, Sep 29, 2008 at 3:42 PM, Brian Webster [EMAIL PROTECTED] wrote: What is the best-recommended resource for searching archives of this mailing list? Thanks for your time ___ -- 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] Aheeva With Asterisk
I stumbled upon this call center software that works with Asterisk calles Aheeva. Does anyone else use it? Thanks. ___ -- 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] cisco VAD and Asterisk recordings
Hi all, I'm experiencing problems with VAD activated on a cisco router doing the bridge between an PBX and de asterisk server. The calls are all rights, but on the recordings the silence from the cisco end point doesn't get recorded, so the audio is completely wrong (the words and phrases from this side are all 'glued' togheter and the other (native SIP) are OK) Anyone experienced problem like this? Gabriel Ortiz ___ -- 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 - Failover System
Steve Edwards wrote: The configuration is trivial for my needs and I find it to be rock solid. I love that you can restart it without interrupting calls in flight. Well, almost. :) If you are using record-routing, it's going to drop transaction state so stateful forwarding for subsequent in-transaction requests isn't going to happen, nor is it going to recognise IDs of calls for which subsequent in-dialog requests must be routed. If you are doing completely loose routing, you might be okay. Either way, as you say, it's not going to cause media to cease in the middle of a call, no. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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 - Failover System
Jai Rangi wrote: Openser? for 120 user? I would not do that. This would be an extra layer to configure, support, maintain and one more layer to debug if things go wrong. Its like spending a Dollar when you can be done with a quarter. (my 2 cents) All depends on how important those 120 users are. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Aheeva With Asterisk
Hello, If you are looking for a list of Call Center software packages that work with Asterisk then take a look here: http://www.voip-info.org/wiki/view/Predictive+dialer There are over 20 now I believe. MATT--- On 10/1/08, broadband Voice [EMAIL PROTECTED] wrote: I stumbled upon this call center software that works with Asterisk calles Aheeva. Does anyone else use it? Thanks. ___ -- 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 - Failover System
I agree that an OpenSER solution on top of Asterisk for a 120 users is massive overkill to say the least. High calls-per-second? Multiple Asterisk servers? Multiple vendors? Advanced LCR? or plans for any of that in the near future? Then I think it makes sense to look at fronting Asterisk with OpenSER for such a small amount of users. Asterisk can do everything you'll need it to do otherwise. - D _ Darren Sessions [EMAIL PROTECTED] http://www.darrensessions.com _ On Oct 1, 2008, at 7:44 PM, Alex Balashov wrote: Jai Rangi wrote: Openser? for 120 user? I would not do that. This would be an extra layer to configure, support, maintain and one more layer to debug if things go wrong. Its like spending a Dollar when you can be done with a quarter. (my 2 cents) All depends on how important those 120 users are. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Aheeva With Asterisk
Matt, I saw them all but interested in one that I can use ATA as the agent. They all seem to be based on softphones, my problem is if the computer goes to sleep, the sofphones will not work. On Wed, Oct 1, 2008 at 10:31 PM, Matt Florell [EMAIL PROTECTED] wrote: Hello, If you are looking for a list of Call Center software packages that work with Asterisk then take a look here: http://www.voip-info.org/wiki/view/Predictive+dialer There are over 20 now I believe. MATT--- On 10/1/08, broadband Voice [EMAIL PROTECTED] wrote: I stumbled upon this call center software that works with Asterisk calles Aheeva. Does anyone else use it? Thanks. ___ -- 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] ATA for large networks
I work at a service provider and we use Audiocode MP-124 (Cisco IAD too). They are reliable and stable. We use them for customers at shopping malls. Regards, Armando -- From: Col Ferguson [EMAIL PROTECTED] Sent: Tuesday, September 30, 2008 11:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ATA for large networks We have one hotel using Xorcom devices. It has 1 32 port FXS bank, and 1 24 port FXS + 8 Port FXO. It works great with all the old analog phones in the motel, over the existing wiring. I haven't tried it with a fax though, but modem usage is very hit and miss. The Xorcom guys are looking into this, as it should work. Apart from that problem though, I'm very happy with the Xorcom boxes. To do the 380 extensions though would require 12 of these boxes, so you'd be using 12 USB connection on a single PC. Cheers, Col - Original Message - From: Loic Didelot [EMAIL PROTECTED] To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, October 01, 2008 12:13 AM Subject: Re: [asterisk-users] ATA for large networks I would use Xorcom devices. Its not realy an ATA but you will have less problems managing an asterisk with a few Xorcoms than many ATA devices. Also you might have Fax devices and modems in your building and here Xorcom is definitively a better choice than ATA devices. Loic. On Mon, 2008-09-29 at 01:06 -0700, Vieri wrote: Hi, I would like to know if someone can suggest a multi-port ATA worth buying (at least 8 ports). I have around 380 analog phones to convert to SIP extensions. So I need quite a few ATAs but they need to be enterprise-grade, ie. they need to be reliable and stable. I bought and have been using 11 Grandstream GXW4008 (8-port FXS ATA) in a production environment and have been experiencing stability and quality issues which are not acceptable in a large company. I chose Grandstream because: - it was a cheap way to start - I thought their products were stable and reliable because I had already heard their brand name So since my experience with 11 Grandstream GXW4008 has been overall negative (I need to reboot the devices too often!), I'd like to know if someone could help me decide what brand/model to buy. I would also need to find these products in Europe (or at least deliverable there). I've been considering a few products but I don't know if they are reliable: TopGate TG8048 (48 FXS) Soundwin S2400 (24 FXS) 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. 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 ___ -- 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] Asterisk custom functions
Hi All, i have centos5 system, i have installed asterisk 1.4 branch. i havedone realtime connection with odbc to pgsql. i have created custom functions in func_odbc.conf, all dsn setup and connection is working fine, but custom functions are not being registered to asterisk. i have given queries to functions and using that functions in dialplan. but it is always gives me function is not registered. can any body explain how to register custom functions in asterisk? Thanks, Max Alex Voip Developer ___ -- 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
