Re: [asterisk-users] mixmonitor extension
On 24-01-14 00:37, Marek Cervenka wrote: can someone confirm that mp3 is unsupported? is patch available? Iirc mp3 was in asterisk-addons in asterisk 1.4 and iirc in later versions of asterisk you can enable format_mp3 in make menuselect. what about patch for Opus? uncle google doesnt know Did you really google? http://lmgtfy.com/?q=asterisk+opus Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is it stable?
On 16-01-14 21:37, Gergely Kiss wrote: Dear List, I'm about to build an Asterisk 11.7 based PBX from scratch for our company. I'm in the middle of the planning phase and it turned out that our VoIP provider prefers H.323 protocol for handling voice calls (while SIP is also supported as plan B). It's SIP everywhere and anyone who requires you, in 2014, to use H.323 should get a clue. Avoid them or at least demand SIP. As I never worked with H.323 channels in Asterisk earlier, I'm not sure if it's stable enough to be used in production. No idea. Maybe someone else with H.323 experience will respond. AFAIK it's a dead-end. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is it stable?
On 17-01-14 01:57, Dan Austin wrote: Patrick Lists wrote: On 16-01-14 21:37, Gergely Kiss wrote: Dear List, I'm about to build an Asterisk 11.7 based PBX from scratch for our company. I'm in the middle of the planning phase and it turned out that our VoIP provider prefers H.323 protocol for handling voice calls (while SIP is also supported as plan B). It's SIP everywhere and anyone who requires you, in 2014, to use H.323 should get a clue. Avoid them or at least demand SIP Bah. There is nothing wrong with a working H.323 stack. Just assuming that they will have a working SIP stack because of the date can lead to heartache. By itself there is nothing wrong with a working H.323 stack. I just would not use it :-) Using H.323 for one provider while any backup or alternative providers probably use SIP results in needing two stacks in testing production. It also requires the admins to gain knowledge of a legacy protocol. Maybe there are some incumbents or service providers with legacy H.323 equipment continuing to offer H.323 service. I get that. But for a business building a VoIP PBX from scratch H.323 does not make sense from a cost and operations point of view. As I never worked with H.323 channels in Asterisk earlier, I'm not sure if it's stable enough to be used in production. No idea. Maybe someone else with H.323 experience will respond. AFAIK it's a dead-end. The ooh323 channel has been fairly reliable in our use case, which involve connecting to a commercial IP PBX with crud SIP support. Only you can tell if it will work for you however, as sadly many times new core features only get tested against the SIP channel(s), or worse only implemented there as well. Our current Asterisk version is 11.5.1 The OP mentioned that his VoIP provider prefers H.323 so it seems to be about trunking. IMHO fairly reliable is not something that is acceptable for trunking phone service. H.323 is what Gopher is to HTTP/webservers. When was the last time you used a Gopher service? Would you today still buy Gopher based service because the service provider prefers it? :-) Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?
Hi Steve, On 15-01-14 18:53, Steve Edwards wrote: On Wed, 15 Jan 2014, Patrick Lists wrote: Would you mind sharing where you get the per country IP ranges from? I confess I 'brute forced' it by entering '/8s' into ARIN's web page and noting if the block had been assigned to a 'foreign' NIC -- not really a reliable and robust methodology, but it worked for me. If it works... :-) A great way to kill time while on hold for customer dis-service. Definitely. If any of the calls lasted more than entering 20 /8s I hope it was to cancel the service. I found another solution: install the geoip kernel module from xtables-addons, install the MaxMind GeoIP country database and add some rules to the iptables config to block a country. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?
Hi Steve, On 14-01-14 10:39, Steven Howes wrote: On 14 Jan 2014, at 02:19, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: Thanks for your feedback Paul. The not having outbound trunks is going to be a challenge. Why? it’s what contexts were invented for. Yes that is indeed what they are for but in the case they find a loophole or exploit a bug then not having outbound trunks is much safer. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?
Hi Steve, On 15-01-14 02:44, Steve Edwards wrote: On Tue, 14 Jan 2014, Patrick Lists wrote: ...I guess I'll cook up some dialplan logic that records IP addresses, keeps track of the amount of failed password attempts etc. and block the offending IP addresses... A few iptables rules can protect you from access from China, North Korea, Iran, Iraq, xxxistan, Russia, Nigeria, and any other country you're not expecting calls from. Eliminate 90% of the problem at the front door and you can focus more clearly on the remaining 10%. Yes that's one of the tricks in my bag. Unfortunately it seems that the IP ranges from ip-deny.com are no longer available and even their website has disappeared. Would you mind sharing where you get the per country IP ranges from? Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?
Hi all, I'm looking into adding the ability to call me at m...@mydomain.org on my Asterisk 11 box. Does anyone have any tips or dialplan snippets to allow this kind of access as securely as possible? Thanks, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?
On 14-01-14 02:36, Paul Belanger wrote: On Mon, Jan 13, 2014 at 9:24 AM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: Hi all, I'm looking into adding the ability to call me at m...@mydomain.org on my Asterisk 11 box. Does anyone have any tips or dialplan snippets to allow this kind of access as securely as possible? Well, if you want anybody to call you, you need to leave it open to the public. Meaning, you can't really secure it. Obviously, don't have any outbound trunks configured on the box so that the only location some could dial would be your extension. Thanks for your feedback Paul. The not having outbound trunks is going to be a challenge. So next to fail2ban I guess I'll cook up some dialplan logic that records IP addresses, keeps track of the amount of failed password attempts etc. and block the offending IP addresses together with max simultaneous outband calls and anything else I can think of to beef up security and limit potential damage. Thanks, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem building dahdi from source
On 01/03/2014 03:56 PM, Jonas Kellens wrote: Hello, I am getting the following error when compiling dahdi : [snip] `/usr/src/dahdi-linux-complete-2.7.0.1+2.7.0.1/linux' make: *** [all] Error 2 I have the right kernel sources installed : [root@sip dahdi-linux-complete-2.7.0.1+2.7.0.1]# uname -a Linux sip 2.6.32-431.1.2.0.1.el6.x86_64 #1 SMP Fri Dec 13 13:06:13 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux So what am I missing ? Try DAHDI 2.8.0.1 http://www.asterisk.org/downloads/dahdi Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding
On 12/15/2013 09:55 PM, CDR wrote: I have had the issue for years. The problem is that Asterisk developers are removed from the business. We desperately need simple way to eliminate transcoding when unnecessary. Transcoding brings a server to its knees. It is a very simple new setting in sip.conf prioritize_matching_codecs=yes Maybe have a look at FreeSWITCH. It's extremely flexible so may offer what you want to do. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
On 12/14/2013 01:29 AM, Martin wrote: If I need to use SIP, from where to get the suitable firmware for these Cisco IP Phones 7942G? Be careful, not all versions of SIP firmware work with asterisk. I do have 8-3-1 (cmterm-7941_7961-sip.8-3-1)here and it works just fine with my 7961. Downloaded somewhere. Version 9.x is broken, SIP only works over TCP. I thought that was fixed in the latest 9.x? Where do u download the SIP firmware usually for your Cisco IP Phones? I have a 7961 and just registered at cisco.com then logged in, did a search and was offered the firmware files for free. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Windows
Probably feeding the trolls but here it goes. On 12/04/2013 04:19 PM, CDR wrote: Digium is 100% lost in the map. If they would come up with a Paid version of Asterisk, one that would use the .NET framework in Windows, something simple to install, they could go public on the product. IIRC Microsoft no longer invests in the .Net framework which makes it a bad idea for a product that would live for up to 10 years. Do you really want to bet your business/company that .Net will be there in 5 to 10 years? Linux has a very steep learning curve. A Windows application that would do exactly the same would be a home run. I find Linux easier than Windows. Installing a package on Linux or Windows is not the issue. How is a simple 'yum install asterisk' any more difficult than double clicking on it in Windows? It's what you do afterwards with the OS and package. Asterisk has a much steeper learning curve than either. It's easy to mess up the config and suffer the consequences if the box is Internet facing. Also, Windows has a terrible reputation when it comes to security. Why would anyone want to use Windows for an Internet facing service? There's a reason that Google, Facebook, Twitter and pretty much the rest of the world are powered by Linux and it's not only because it's cheaper. Just because you find Windows easier does not make it a good idea. Note: I am a Linux expert user, but it took me years to get here. And still, moving from regular RHEL 6.0 to Fedora 20 (RHEL 7) is a pain in the neck. There is probably a saying about people calling themselves experts and then complain about a move from EL6 to F20 which is puzzling by itself. The .NET framework and Windows server 2012 are miles away in terms of friendliness and on equal footing on performance. I have yet to see a large Telco or ITSP deploy their services on Windows. A while back I have seen some attempts. It was hilarious to hear that the servers had to be restarted every few hours. Performance totally sucked, components would crash and the solution was, even by telco standards, ridiculously expensive. So no, they are not on equal footing when it comes to performance (and other aspects). I don´t mean another slow cygwin port, I man a native Asterisk for windows. In fact, I would invest on the project if somebody wants to do it. If you really want to use Windows then have a look at FreeSWITCH as it's available on Windows too. Then there is also Lync and 3CX. Good luck keeping your Windows boxes from getting hacked with all the financial and other damage it would cause. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi-tools 2.8.0-rc4 - udev rules not installed?
On 12/03/2013 06:35 PM, Russ Meyerriecks wrote: This is why we love release candidate feedback! Thanks! I've managed to mis-tag rc4 and missed all of Oron's commits. Cutting a v2.7.0.2 and a (correct) v2.8.0-rc5 today. Thanks. I'll give rc5 a spin when it arrives and report back if I find anything else out of the ordinary. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems compiling dahdi modules
On 12/02/2013 04:19 PM, Russ Meyerriecks wrote: This is fixed on the dahdi-linux master branch and will be included in the next release: More info: http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=summary http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=5ec9d756aac1a0eb5c1f48eb110e80946b43f41a https://issues.asterisk.org/jira/browse/DAHLIN-330 Thanks Russ. Any chance there will be a v2.8.0-rc3 release with these fixes? Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI 2.7.0.1 and CentOS 6.5
On 12/02/2013 10:09 PM, Bakko wrote: Hello, during DAHDI 2.7.0.1 compilation on CentOS 6.5 64bit, I have this error: [snip] This was discussed earlier today and Russ pointed to the fixes: http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=summary http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=5ec9d756aac1a0eb5c1f48eb110e80946b43f41a https://issues.asterisk.org/jira/browse/DAHLIN-330 The fix will be in 2.8.0-rc3. Either wait for the rc3 or add the patch to your build (don't know if it works on 2.7.0.1). Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi-tools 2.8.0-rc4 - udev rules not installed?
Hi, I just looked at 2.8.0-rc4 and noticed the udev rules/apps change which are now supposed to be part dahdi-tools. After make, make install and make config it seems the dahdi.rules are not installed. I couldn't find a reference to it in the Makefile either. Did I miss something or has the move to dahdi-linux not yet been completed? Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail greeting playback issues?
On 11/26/2013 12:24 AM, Doug Lytle wrote: Bryant Zimmerman wrote: Hey all I believe I found the bug in Asterisk 11.xxx If someone can help me verify it. Actually, I wouldn't consider it a bug. I've know for years that you need to answer a channel before you play back audio or strange things can and will happen. That's what I do since the 0.x days. IIRC in recent Asterisk versions some apps answer before doing anything else. Guess the voicemail app is not one of them. I always answer first followed by a small Wait and then execute the app. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?
On 10/28/2013 07:29 PM, Eddie Mikell wrote: All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine That's a good start. Now what have you done to conclude that the Asterisk server is not the cause of your problems? More than enough bandwidth That's irrelevant. It's about the quality of that bandwidth. Have you figured out if there might be a lot of packetloss or are you perhaps on a cablelink which is a *shared* medium? Once your link hits the box in the street it shares it with others who might be eating up all the bandwidth with their torrent downloads etc.? Use tools like iperf, smoke ping and mtr to see if there are obvious problems on the route to your VoIP provider. Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure what option would be the best Once the packets leave your premises and your ISP/cable company starts messing with them a QoS setting is generally not honored so not very helpful unless your LAN is congested. Put analog lines in the conference room to avoid the dropouts - leave the sip lines in place for day to day use If those analog lines are cheap, easy to get then as an intermediate solution I would order those analog lines as fast as I could. Or fix the VoIP problems, whichever is faster. Hire a consultant An experienced VoIP consultant should be able to tell you what is or could be causing your problems. With your users sick of phone service it suprises me that you haven't already hired one. Ditch the system and buy a pre-packaged system - RingCentral or some such. And what if it's your Internet link or the route to your VoIP provider? What if your VoIP provider is messing up? There are no local asterisk professionals who can help, and we are a little leery of opening up our system to outside consultants. If you don't want that then you don't want that but given the state your users are in I would be less worried about giving a Consultant access to the Asterisk box and more worried about my job :-) Anyone else face the above, and finally abandoned Asterisk for a commercial system? I have seen that once years ago where some clueless sales guy had totally oversold an ancient Asterisk/Bristuff/ISDN setup which was very buggy and crash prone. There was no way to make that work reliably. After the supplier failed for months I was brought in to review the setup and possibly fix it. Told the customer to cut its losses. So they kicked out their supplier and opted for a different setup. We have 167 users. I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the conference rooms. I don't know how Grandstream is these days. I thought the GXP2100 was ok but I guess you already know if there's a problem with those phones from the (lack of) intra-office call complaints from your users. Suggestions welcome. Hire a Consultant or someone who has been part of this Community for a while and is well known on this list or in #asterisk on irc. Provide remote access if required. Change passwords afterwards. If you really don't want to provide remote access then find a reputable VoIP provider with a switch physically as close as possible to your location, get a DID for a few bucks, hook it up to your Asterisk box and route it to a line on your phone, grab your cell, call that DID and see if you still have the problem. It wouldn't be the first time that the link between you and your VoIP provider just doesn't cut it. Or maybe your VoIP provider just sucks and you need to change to a different one. Both flowroute.com and voip.ms work well for me (no affiliation). Or maybe your Internet link sucks and you need to change your ISP. Good luck. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk TON number
On 09/25/2013 09:22 AM, Endri Stefani wrote: Hi Greeting to all you out there. I am new at asterisk, I have been working with PLMN platforms telecommunication for 5 years with NSN and Huawei. We have recently built an asterisk PBX with Trixbox and connected it to our MSS using Digium E1 cards(ISDN). Everything went smoothly as there are tons of information out there, except for the TON number. If you have worked in Telecommunication you will know the importance of TON flexibility. All the posts online suggested to update under Chan_dahdi.conf: pridialplan = international prilocaldialplan = international or other TON value ,restart the platform and then trixbox1*CLI dialplan reload I have already done this with no success. Are there other changes I have to make in order to change dialplan? Afaik the Trixbox Community Edition is no longer developed. So unless you use a commercial Trixbox you are perhaps better off with just the latest stable Asterisk (asterisk 11.5.1 or 11.6.0-rc1, dahdi 2.7.0.1, libpri 1.4.14, libss7 1.0.2) or if you need a GUI have a look at Elastix, the FreePBX distro, etc. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk TON number
On 09/25/2013 01:57 PM, Endri Stefani wrote: Hi Patrick, If I use latest stable asterisk will I be able to change dialplan by changing pridialplan in chan_dahdi.conf? AFAIK yes. You may also want to check out Asterisk The Definitive Guide (4th edition is the latest). Paperback version: http://www.amazon.co.uk/Asterisk-Definitive-Guide-Russell-Bryant/dp/1449332420 And voip-info.org has a ton of information (not always current though): Search pridialplan on this page for more info: http://www.voip-info.org/wiki/view/chan_dahdi.conf Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Somewhat-OT: Stupid NAT tricks to learn from Apple?
Hi Kristian, On 09/20/2013 03:17 PM, Kristian Kielhofner wrote: I've been spending some time looking at some of the significant changes Apple has made to Facetime in iOS 7. I'm far from an Apple fanboy but some of them are pretty interesting: - multiplexing everything over a single UDP port - deflate compression with SIP - various /slight/ protocol violations ;) More here: http://blog.krisk.org/2013/09/apples-new-facetime-sip-perspective.html As SDP bodies swell more and more can we hope to build significant support for multiplexing and deflate compression in the SIP-focused open source ecosystem? Thanks for sharing your analysis. Interesting read. Makes me wonder why not more vendors/projects are doing port multiplexing. Let's hope it will pick up steam now that Apple has implemented it. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN outgoing caller id
On 08/27/2013 08:04 PM, Gergo Csibra wrote: Hi, is anybody out there who can set the outgoing caller id on ISDN (CAPI or misdn) channels? I've tryed everything what I found in forums, os voip-info.com but no luck. I use a fritz card with CAPI in my first installation (1 BRI), and a hfc 4 port bri card with misdn on other. The first installation have p-t-mp configuration, the second one is p-t-p. Both configuration is EuroISDN in Hungary. So, can anybody help me? Have you checked with your Telco if they allow you to change the callerid? If yes, are you setting the callerid to a number that you are allowed to use? You can't just set callerid to any number you like. You must own the number which you want to set callerid to. I have no problem setting the callerid on outgoing calls via chan_capi to one of the numbers that the telco assigned to me. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Am I being hacked?
On 08/19/2013 08:10 PM, Eric Wieling wrote: One of Asterisk's dirty little secrets is that it does not show the source IP when a device or hacker tries sending a call without registering. The rejection message in the logs do not show the IP of the attacker. Yes it sucks, yes it has been that way for many many years. Are you aware of a patch that would show the source IP in the console and logs? Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Am I being hacked?
On 08/19/2013 08:55 PM, Steve Edwards wrote: On Mon, 19 Aug 2013, Ira wrote: [2013-08-18 05:56:29] NOTICE[17089][C-00a8] chan_sip.c: Failed to authenticate device 390sip:3...@xx.xx.xxx.xxx;tag=2762c06e xx.xx.xxx.xxx is my public I.P. What kind of filtering are you doing? Iptables? Rather than playing 'wack-a-mole' with hackers, my first line of defense is to 'white-list' just the few legitimate connections between my clients and their SIP providers. If your situation requires remote and mobile access, can you at least 'black-list' certain countries with a propensity for hacking? Do you need access from China, North Korea, Iran, etc? You can eliminate a very large percentage of hacking attempts with just a few rules. Then you can focus better on the remaining threats. Agree. The ip blocks from ipdeny.com come in handy either blocking countries that have no business accessing your Asterisk box or whitelisting countries/ip ranges that do. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Am I being hacked?
On 08/19/2013 09:29 PM, Eric Wieling wrote: Actually, you can try enabling the security logging destination in logger.conf. I believe that may contain the info, but it is new in Asterisk 11. 1.8 and earlier does not have this. Thanks I'll give that a try. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo Cancellation
On 07/25/2013 11:51 AM, bilal ghayyad wrote: Hello; If our Digium Telephony Card does not support echo cancellation like (1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome the echo? Use the free OSLEC echo canceller software module or Digium's commercial HPEC echo canceller software module. Google is your friend. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LUA
On 07/18/2013 03:56 PM, jacob.e.mi...@l-3com.com wrote: I am attempting to setup my server to use Lua for the dialplan (extentions.lua), but I am unable to get the asterisk configure script to find the installation of Lua on my box. I have downloaded the Lua sources from the www.lua.org site, and I have installed via the “make linux install” command. I can execute lua scripts via the command line, but asterisk configure script is unable to find the installation of Lua. That's probably because Asterisk is not looking in /usr/local. I am on a closed network, so no access to the internet so I am not able to just install Lua using yum. You should have downloaded the lua RPMs to e.g. your laptop, then copy them to your Asterisk box with e.g. a USB stick and then install the Lua RPMs on your Asterisk box with: $ sudo yum install ./lua* You can find the CentOS 6.4 x86_64 Lua RPMs here: http://mirror.stanford.edu/yum/pub/centos/6.4/os/x86_64/Packages/ Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adjusting confbridge call quality
On 07/10/2013 06:46 PM, Chris Gentle wrote: [snip] and then others can connect via SIP. For some reason, when the speaker says words with S's and F's, they almost sound distorted. Not quite static but you can tell the quality has been affected. May just be a side-effect of 8,000 Hz. Just wondered if there way some way to improve that. The distorted S and F are prevented by a pop filter in front of the mic. Are you using a pop filter? Also if you are using a cheap mic, do yourself a favor and invest in a decent mic. It will make a world of difference. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11 security log, fail2ban, drive-by SIP attacks
On 07/08/2013 01:46 PM, Giles Coochey wrote: Just a note that I did a little work to extend FreePBX distro with some extra Fail2Ban which deals with some drive-by SIP registration attempts. My regex is poor to middling, but the steps detailed here: http://www.coochey.net/?p=61 manage to stop IPs which try to authenticate against Asterisk which FreePBX were not able to stop before. I would welcome any improvements anyone would care to submit and I'll extend the article a little. The changes need the Asterisk security log feature, which I think was only introduced in later versions of Asterisk (e.g. v11). It seems your rule is not yet present in fail2ban 0.8.10.0. The only one close to it is: SECURITY%(__pid_re)s [^:]+: SecurityEvent=InvalidAccountID,EventTV=[0-9-]+,Severity=[a-zA-Z]+,Service=[a-zA-Z]+,EventVersion=[0-9]+,AccountID=[0-9]+,SessionID=0x[0-9a-f]+,LocalAddress=IPV[46]/(UD|TC)P/[0-9a-fA-F:.]+/[0-9]+,RemoteAddress=IPV[46]/(UD|TC)P/HOST/[0-9]+$ See https://github.com/fail2ban/fail2ban/blob/0.8.10/config/filter.d/asterisk.conf Might be an idea to submit it for future inclusion. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is it possible to provision lock Aastra phones?
On 07/06/2013 03:35 PM, Bruce B wrote: Thanks Patrick. Do the encrypted config files safe guard against hard resets such as Web Recovery mode - aka holding down 1 # sign at startup? My main purpose is to lock the sets due to contract terms so I'd rather not see user steal the phone and break contract without payment. It's been a quite a while since I setup Aastra provisioning so a bit fuzzy on the details but from what I recall it worked pretty well. The Admin Guide can probably give you more info. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is it possible to provision lock Aastra phones?
On 07/06/2013 08:15 AM, Bruce B wrote: Hi everyone; Is it possible to provision lock Aastra phones to provider so that no soft or hard reset can unlock them? Iirc you can use encrypted configs using an app called anacrypt and lock them down. The admin guide (3.2.2) has more details in section 2-14, 5-44 - 5-46 and A-187 - A-189. http://www.aastra.com/cps/rde/aareddownload?file_id=6950-16962-_P06_XMLdsproject=aastramtype=pdf http://www.aastra.com/document-library.htm?curr_nav=2curr_fam=Aastra+6750iprod_id=6950# Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Analog card and Asterisk
On 07/04/2013 05:32 PM, 杨华杰 wrote: Hi I just bought some digium analog cards and I would like to build an IVR system for my customers. However I am googling and googling , I didn't find any blog and instruction for beginners like me. So I come here for help. Any tips or blogs will help. http://www.asteriskdocs.org/ https://wiki.asterisk.org/wiki/dashboard.action http://www.asterisk.org/community/documentation Have fun! Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why does it take several seconds to interpret DTMF-input ?
On 06/11/2013 04:44 PM, Jonas Kellens wrote: [snip] Ok thanks. Any idea how I can resolve this ? Even if there *can* be more than 1 digit, in case there is only 1 digit it should go faster. Would it help if they pressed for example 1 followed by the # key? If not then, as Eric mentioned, redesign your dialplan. Any IVR with a double digit amount of options needs some rethinking. IMHO the average attention span of a person is such that at option 6 they forgot options 1 through 5. And if the option explanations last longer than 5 seconds it gets even worse. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OC3/STM-1 Line Card
On 06/09/2013 06:35 PM, Nick Khamis wrote: Anyone? Sangoma has a multiplexer: http://www.sangoma.com/products/stm1mux-fiber-multiplexer/ Which you could then use with: http://www.sangoma.com/products/a116-16-span-t1e1j1-board/ And there is this card: http://www.signalogic.com/index.pl?page=asterisk_ip_pbx Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which dahdi/libpri combo for BRI/PtmP ?
On 06/06/2013 05:55 PM, Olivier wrote: Hi, I need to rebuild a system which has 4 BRI ports and is connected to Point-to-multiPoint lines, in a country where telco often drop lines for energy savings. I think the dropped D-channel issue should be handled by a very recent DAHDI. If there are still issues file a bug. Don't know about PtMP. I'm planning to use latest 11.4.0 asterisk version along with dahdi and libpri (no misdn). Which version are recommended for Dahdi and Libpri ? Probably the very latest you can find. If the latest stable does not work for you then try an RC of an upcoming version or get master from git. My main requirements are, beside having calls coming in and out: - keep a meaningful pri show spans status (pri show spans outputs Down when the line is down (cable unplugged, no q921 traffic, ...)) - avoid sporadic ERROR messages in logs. Guess you have to test. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe exit status?
On 06/03/2013 06:47 PM, Matthew Jordan wrote: On 06/02/2013 08:36 PM, Patrick Lists wrote: Hi, Does MeetMe in Asterisk 11.4 set some kind of exit status or a var so I know for example if a conf ended normally or if someone gave a wrong conf number or pin? Thanks, Patrick There is no channel variable that provides that level of granularity. The closest available is the MEETMESECS channel variable, which tells you how many seconds the participant was in the conference. You can find a full list on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/MeetMe+Channel+Variables Thanks Matt. I'll see if I can use MEETMESECS. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Missing dahdi-firmware RPM in asterisk-current repo at http://packages.asterisk.org
Hi, The dahdi-firmware package seems to be missing in the asterisk-current repo on http://packages.asterisk.org -- Finished Dependency Resolution Error: Package: dahdi-linux-2.6.2-1_centos6.x86_64 (asterisk-current) Requires: dahdi-firmware Can this please be fixed. Thanks, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe exit status?
Hi, Does MeetMe in Asterisk 11.4 set some kind of exit status or a var so I know for example if a conf ended normally or if someone gave a wrong conf number or pin? Thanks, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi driver not getting install
On 05/13/2013 01:14 PM, Salaheddine Elharit wrote: hi You can download a tarball of the release here: http://downloads.asterisk.org/pub/telephony/dahdi-linux/dahdi-linux-2.6.2-rc1.tar.gz At least give the link to the latest release which is not 2.6.2-rc1 but 2.6.3-rc1: http://downloads.asterisk.org/pub/telephony/dahdi-linux/dahdi-linux-2.6.3-rc1.tar.gz Or he could use the yum repo from Digium: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages http://packages.asterisk.org/centos/6/current/x86_64/RPMS/ Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones
Hi Carlos, On 04/28/2013 10:56 PM, Carlos Alvarez wrote: We have a new customer with a lot of old phones like the 9133i. They won't register, and we see some very strange behavior with them. If the SIP peer exists, they simply fail silently, with no error in the CLI or the messages log. Nothing works, but no errors. If the peer does not exist, it's clear that it's registering improperly: [2013-04-28 13:34:31] NOTICE[3058] chan_sip.c: Registration from 'abc123 sip:abc123@' failed for '68.2.x.x' - No matching peer found Typically of course we'd expect to see: sip:abc123@server We're running the latest available firmware, but it's from 2009. Any ideas on this before we just trash all the older phones? I reviewed one of those a long time ago. I'm afraid all I can remember is that it had its fair share of issues. I did a lot of factory resets and had to set the config through tftp *or* the UI but not mix both. Did you try doing factory resets (page 9-11 in admin guide) and/or downgrading to an older firmware release? Main info page: http://www.aastra.ca/document-library.htm?curr_fam=Aastra+9000icurr_nav=2prod_id=6441 The 1.4.0 firmware: https://www.voipon.co.uk/products/download/aastra_9133i.zip The 1.4.3 firmware: http://www.aastra.ca/cps/rde/aareddownload?file_id=6441-6449-_P07_XMLdsproject=www-aastratelecom-commtype=zip Release notes of 1.4.3: http://www.aastra.ca/cps/rde/aareddownload?file_id=6441-6427-_P07_XMLdsproject=www-aastratelecom-commtype=pdf Manual http://www.aastra.ca/cps/rde/aareddownload?file_id=6441-6425-_P07_XMLdsproject=www-aastratelecom-commtype=pdf Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax - sound/tone - dealing with SPAM
On 04/04/2013 09:54 PM, Joseph wrote: +1.7044972383 If that number is his actual number, maybe create a script that calls him 10 times an hour, every hour between 00:00 - 07:00am and plays screaming monkeys every time he picks up (or his voicemail kicks in). Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TigerJet 320G Chip / TDM400 Chipset / DAHDI Support
On 04/03/2013 02:48 PM, Marshall Henderson wrote: Hi Tzafrir- I know where to find the DAHDI source, but I was more asking where to actually find which chipsets are supported within the source. Any thoughts? Have you checked the PCI IDs in the source? Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TigerJet 320G Chip / TDM400 Chipset / DAHDI Support
On 04/03/2013 08:34 PM, Marshall Henderson wrote:
Hi Patrick- Yes, I did find the list of PCI IDs (I think). Do these look
right (from wctdm.c):
static DEFINE_PCI_DEVICE_TABLE(wctdm_pci_tbl) = {
{ 0xe159, 0x0001, 0xa159, PCI_ANY_ID, 0, 0, (unsigned long)
wctdm },
{ 0xe159, 0x0001, 0xe159, PCI_ANY_ID, 0, 0, (unsigned long)
wctdm },
{ 0xe159, 0x0001, 0xb100, PCI_ANY_ID, 0, 0, (unsigned long)
wctdme },
{ 0xe159, 0x0001, 0xb1d9, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmi },
{ 0xe159, 0x0001, 0xb118, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmi },
{ 0xe159, 0x0001, 0xb119, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmi },
{ 0xe159, 0x0001, 0xa9fd, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmh },
{ 0xe159, 0x0001, 0xa8fd, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmh },
{ 0xe159, 0x0001, 0xa800, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmh },
{ 0xe159, 0x0001, 0xa801, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmh },
{ 0xe159, 0x0001, 0xa908, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmh },
{ 0xe159, 0x0001, 0xa901, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmh },
#ifdef TDM_REVH_MATCHALL
{ 0xe159, 0x0001, PCI_ANY_ID, PCI_ANY_ID, 0, 0, (unsigned long)
wctdmh },
#endif
{ 0 }
};
So, next question, how do I take those device IDs and find the
associated chip?
Look up the PCI ID at for example:
http://pciids.sourceforge.net/
http://www.pcidatabase.com/
The 0x159 is the TigerJet chip. The third column is the vendor. So if
you lookup 0xb100 you will find that it is OpenVox.
Going a step further, after a bit of research it appears some
manufacturers have gone away from dedicated hardware chips like the
TigerJet 320G to FPGA general purpose chips with firmware to control
operation. Is this correct? Any thoughts?
No idea so no thoughts :-)
Regards,
Patrick
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Re: [asterisk-users] ERROR: Unknown signalling method ss7
On 03/14/2013 11:04 PM, mohsen feyzzadeh wrote: Hi all I installed DAHDI Version - 2.6.1 DAHDI Tools Version - 2.6.1 libss7-trunk Asterisk 11.0.1 from source on Fedora 12 x86_64. In case the 12 in Fedora 12 was not a typo, you do realize that Fedora 12 has been end-of-line for years and has more security holes than Swiss cheese? It makes sense to upgrade to the latest version of Fedora (which is 18) or switch to CentOS 6.4 which is more suited for server applications. You may also want to look at the latest versions of DAHDI (2.6.2/2.6.3rc) and Asterisk (11.2.1) assuming both work with an appropriate version of libss7. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] digium card and virualbox
On 03/11/2013 04:18 AM, bilal ghayyad wrote: I am not mixing. I need this for LAB testing. How? This PCI passthrough, how to enable it on virualbox? It's in the VirtualBox manual. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Laptop error
On 03/11/2013 12:53 PM, termo termosel wrote: Hi, I have Ubuntu and Asterisk 11.2.1 in a boot USB. When I put it in desktop computer, asterisk starts without problem but if I insert the same USB in a laptop computer Asterisk doesn't start. Is it possible because different microprocessors? Yes. If you made the USB stick on a x86_64 (64 bit) computer and then try it on a x86 (32 bit) laptop, it will not work. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS from asterisk
On 03/11/2013 07:07 PM, Asghar Mohammad wrote: HI Bilal, i am using chan_mobile for call termination, you can use it but you need to tweak chan_mobile.c it is broken from a long time. let me know if you want give it a try. If you could send the patches you made to chan_mobile to this mailing list then other Asterisk users can benefit from your work and use chan_mobile too. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Serviced Office operator panel
On 03/12/2013 12:07 AM, Andrew Yager wrote: Hi, I'm trying to find (with some desperation now) a decent web based or application based UI that integrates with an Asterisk based PBX and is designed for a Serviced Office environment. Key features we're looking for: Don't know if it covers your requirements but here's another commercial solution: http://www.getisymphony.com/ Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where can get the latest manual our user guide
On 02/08/2013 06:35 AM, Ding Peng wrote: Hi, everybody, Where can I get the manual or user guide of latest asterisk version, 1.11.x? I want to know the syntax and usage of all the supported functions or something like that in the latest version. You can find one on the O'Reilly website. Don't recall the link so you have to google for it. And the Asterisk wiki has a lot of info about version 11. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DECT Solution
On 01/24/2013 10:37 AM, Zyumbilev, Peter wrote: Hello, I need to setup system of aroud 60 DECT phones with asterisk. So far I found http://www.grandstream.com/index.php/products/ip-voice-telephony/enterprise-ip-phones/dp715_710 However is there some cheap base station(similar to GSM cell) so I can handle all DECT phones centralized and plug it inside asterisk ? Aastra has DECT base stations that can hook up to an Asterisk server. Last time I set one up it worked fine. You may want to try out several different brands of DECT phones and see which one the users like best. You don't want to get 50 support calls a day from your users complaining about how much the DECT phones suck. http://www.aastra.com/product-families.htm?curr_cat=DECT+Infrastructurecurr_type=Familymode_f=1mode_c=1mode_l=4 Polycom also has DECT stuff. I doubt it will come cheap. http://spectralink.polycom.com/dect_communications/index.html Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] clicking sound with alaw codec
On 01/24/2013 09:44 PM, Richard Kenner wrote: [snip] When I use alaw, the path from Asterisk to the Alcatel is completely clean, but the other way has a set of clicks that kind of sound like old-fashioned audio noise. [snip] It's been ages since I experienced that but things to check that come to mind in no particular order are: - DAHDI settings (sync source) - Asterisk server not properly grounded - timing is off (check logs) - shared interrupts (make sure nic/TDM card have their own) - jitterbuffer settings (try on/off) - echo cancellation going bonkers (OSLEC?) - QoS (proper priority for voice packets?) - PCI slot (if you have a card, try changing the slot it's in) Use Wireshark to see the difference between a good call and a bad one. If you see a lot of time jumps on the bad call then look at your network/QoS. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] clicking sound with alaw codec
On 01/24/2013 11:57 PM, Richard Kenner wrote: - jitterbuffer settings (try on/off) I added jbenable=yes and get lots of: [Jan 24 17:53:41] WARNING[12317]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-6c7 received frame with invalid timing info: has_timing_info=1, len=0, ts=371371424, src=RTP [Jan 24 17:53:41] WARNING[12317]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-6c7 received frame with invalid timing info: has_timing_info=1, len=0, ts=371371434, src=RTP Check https://issues.asterisk.org/jira/browse/ASTERISK-12042 Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Details process to configure Asterisk in CENTOS
On 01/22/2013 08:54 AM, Sakharam Thorat wrote: Can anybody send me Detailed process to configure Asterisk in CENTOS ?? Detailed description highly appreciated. Start by reading the Asterisk book, check asterisk.org and Google around to see if your question has already been answered. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need Help
On 01/17/2013 09:05 PM, Joe Ruffolo wrote: Hi all! In need of some serious help. We currently run Trixbox and Cent Os on a 2u server for our small business’s phones system. Afaik Trixbox is no longer maintained and their forum are hardly active anymore so it may be a bit of a challenge to get support. If you really need a GUI like Trixbox then I suggest you have a look at Elastix which is very much alive and has a large community and professional services to help you out. See http://www.elastix.org/ Or have a look at Digium's Switchvox (payware). Whatever you do/choose, make sure that your box is secure if you open it up to the Internet. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
On 01/02/2013 07:11 PM, Carlos Alvarez wrote: The number of questions posted here that are easily answered with a search or which are far too basic and open (how do I make Asterisk work) is very high these days, and that does kill a list. A lot of us are interested in helping people who help themselves, and solving complex problems. I've seen many tech lists die off when people stop trying to help themselves and ask intelligent questions. Good point Carlos and I share your feeling. On the Postfix mailing list, when someone asks a basic how do I ... question, inevitably the response is one or more links to a section in the documentation. And that works really well. The interesting problems discussed on that ML outnumber the questions from those who can't be bothered to try to help themselves by spending a couple of minutes reading the docs. I would welcome similar responses on this mailing list to improve the S/N ratio. As to top-posting, another example of when I think it's generally acceptable is people using tablets. I have found no way on either my iOS or Android tablets to quickly/easily post in the traditional manner. If I'm faced with spending a few minutes carefully trimming a useful reply or just not posting it at all, I'm likely to choose the latter if I'm on a list that says absolutely never top post. I only use Thunderbird to post but I now have seen several arguments that MUAs like Outlook and iOS/Android clients are simply not capable of bottom posting trimming. Perhaps the list admins could take that into account when appropriate. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
On 01/02/2013 06:20 PM, Steve Totaro wrote: I became a list member way before any such rule and never had to click through and agree to these update ToS. I am grandfathered in. Just looked it up. I see my first post back in April 2003, yours in September 2003 and Jon in March 2003. Wow you find something fun to play with and suddenly a decade has passed :-) Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
On 01/02/2013 09:46 PM, jon pounder wrote: On 01/02/2013 03:22 PM, Patrick Lists wrote: On 01/02/2013 06:20 PM, Steve Totaro wrote: I became a list member way before any such rule and never had to click through and agree to these update ToS. I am grandfathered in. Just looked it up. I see my first post back in April 2003, yours in September 2003 and Jon in March 2003. Wow you find something fun to play with and suddenly a decade has passed :-) Are you sure about that ? I know I was doing stuff with asterisk back in the LSS days and that was around 2001 I only looked at the list archives. LSS definitely predates anything else so it's safe to say you are dinosaured in :-) http://lists.digium.com/mailman/listinfo/asterisk-users Here's to another decade of fun! Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
On 12/30/2012 04:26 PM, Ron Wheeler wrote: I participate in a lot of lists and top posting is now the norm since people want to see quickly if the message is worth reading. Isn't it a bit of a stretch to extrapolate your experience with your lists to top posting being the norm? I am subscribed to several lists and bottom posting, proper trimming and commenting inline is the norm there. Actually the norm is determined by the list rules. If the list rules say one must use bottom posting then one should use bottom posting. If someone does not like that then don't subscribe, find another source to ask a question (the forum, LUG, hire a consultant) or just bottom post. Questions come before answers. Answers come after questions. -1 against changing rule #5. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk crashing when trying to start a Jabber session with ejabberd
On 11/13/2012 12:11 AM, Phil Reynolds wrote: [snip] It turns out to be a known issue: https://issues.asterisk.org/jira/browse/ASTERISK-19532 ... and can be fixed by applying the patch at: https://issues.asterisk.org/jira/secure/attachment/43441/xmpp_no_crash_with_ejabberd.patch I will file the details with Debian too... Is it an omission that this fix has not been applied to the 11 tree? From the looks of ASTERISK-19532 it seems that the fix has only been applied to 1.8 and 10. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk crashing when trying to start a Jabber session with ejabberd
On 11/13/2012 07:05 PM, Michael L. Young wrote: [snip] Is it an omission that this fix has not been applied to the 11 tree? From the looks of ASTERISK-19532 it seems that the fix has only been applied to 1.8 and 10. If you click on the link for ASTERISK-19532, there is a tab in the Activity section labeled Subversion. It shows that the patch was applied to 1.8, 10, 11 and trunk. Thanks Michael. Missed that one. Good to know. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] B200p card - use dahdi or mISDN?
On 10/16/2012 08:50 AM, Sebastian Arcus wrote: I've just bought an OpenVOX B200p ISDN card - and if I remember correctly from last time I used one of these - it is possible to use either DAHDI or mISDN with it. Are there any factors to consider when choosing which software to use? Is one better than the other - or does one have features which are not present in the other? I would go for DAHDI so you can use the card like you would use any Digium card. OpenVOX also seems to focus on DAHDI integration. Looking at the OpenVOX site it seems that you will need to use the patched DAHDI from here: http://downloads.openvox.cn/pub/drivers/dahdi-linux-complete/ Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk installation under a single directory
On 10/15/2012 09:07 AM, sudeep melekar wrote: [snip] i m completely new to asterisk so any help would be appreciated If you are totally new to Asterisk I recommend you first read the Asterisk book and go through the wiki. Both have sections how to install the various Asterisk components. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting Skype to Asterisk
On 10/12/2012 11:17 PM, Philip Bennefall wrote: From what I gather, it costs extra for each channel even for direct Skype to Asterisk calls. Since my plan was to use this for business purposes, I'd need at least something like 30 channels which would be way out of my monthly budget unfortunately. If you *really* need this then have a look at FreeSWITCH which has a module for Skype calls (in/out) without the need for Skype Connect and its fees. Afaik you can use a regular Skype account and iirc even multiple Skype accounts. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LDAP Driver and VoiceMail
On 10/04/2012 10:00 PM, Phil Daws wrote: Hello: I am investigating the possibility of using LDAP for storing certain Asterisk configuration parameters. I have examined res_ldap.conf and see where mailbox can be defined from AstAccountMailbox but I do not see where the password can be stored ? I've never looked at res_ldap but wouldn't a look at the schema tell you that? Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Unanswered calls
On 10/05/2012 11:51 AM, Shanavaz E A wrote: Hi, No replies until now. Some one please help... There must be some people who are using it... Thanks No idea but since Asterisk is making you money why don't you hire an experienced Asterisk consultant to get it resolved. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to log caller IP address in the CDR?
On 10/05/2012 02:10 PM, Benoit Panizzon wrote: Hello We had this situation: Some bot-net did try to guess SIP logins and finally succeeded. The Asterisk Server was abused to call a large number of expensive destinations. I'm sorry to hear that. In the Asterisk source there is a doc that focuses on security. you might want to read that. Google should give you more information about Asterisk/SIP security. Also you may want to install something like fail2ban which prevents brute forcing by banning originating IP addresses after a few failed attempts. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX via Asterisk
On 09/27/2012 08:15 AM, Shanavaz E A wrote: [snip] Patrick, can you please give the steps to configure fax with iaxmodem and hylafax. Is it free to use? It's been years since I set it up so I don't know exactly how to configure it anymore. But I do remember that I found some howto/docs via Google so try that. And yes both Hylafax and iaxmodem are free. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 100 outgoing calls - 10 at a time - /var/spool/asterisk/outgoing/
On 09/28/2012 03:01 AM, Patrick Archibald wrote: Hi, Is there a way to move 100 .call files in to /var/spool/asterisk/outgoing/ at once and have Asterisk call at maximum 10 at a time? Afaik that is not possible. Wouldn't it make more sense to move call files in batches of 10 to outgoing/? Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX via Asterisk
On 09/26/2012 05:53 PM, Mark Robinson wrote: Hello. I have asterisk 1.8.18 which connects to ISDN PRI. All phones are sip, Aastra 6757i. Everything works as expected. We also have a FAX machine. We need to be able to use that FAX machine to send or receive faxes. We are planning to have a dedicated did for faxes. Before, FAX machine was connected directly to pots line. Any digestions how to accomplish it? In addition to Markus' suggestions you could also look at using iaxmodem and Hylafax. I've been using that for years and it works great. Once a fax is received it is emailed to whatever you configure. Sending faxes works too in libreOffice. If you have a bunch of DIDs you could give people their own direct line and fax number. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR
On 09/25/2012 11:18 PM, Logan Bibby wrote: MyISAM would be best, in my opinion. The features that cause the little bit of performance overhead in InnoDB wouldn't be necessary for CDR storage. Iirc InnoDB is ACID compliant so might be preferable if MyISAM is not. More information here: http://en.wikipedia.org/wiki/ACID https://blogs.oracle.com/MySQL/entry/comparing_innodb_to_myisam_performance Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium AEX410, MTNL Mumbai Caller-ID problems
On 09/14/2012 05:26 AM, Raj Mathur (राज माथुर) wrote: [snip] Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express) Your DAHDI and Asterisk versions are old so for starters I would update everything to the latest releases. See asterisk.org. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message
On 09/14/2012 10:25 AM, A J Stiles wrote: [snip] It could be nothing more than a dry solder joint on one of the RJ45s. For the sake of five minutes' work with a soldering iron, that's got to be worth eliminating. Wouldn't that void your warranty? I would take it up with Digium support and let them sort it out. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream VoIP phones
On 01-09-12 04:14, Vladimir Mikhelson wrote: [snip] * Ability to send host name or other CN not equal to the phone IP in TLS negotiation Afaik you usually put alternative CNs in SubjectAltName in the certificate. Have you tried that? Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2
On 08/30/2012 09:45 AM, Gopalakrishnan N wrote: Hi, I have installed libpri, Dahdi 2.6 and Asterisk 1.8.15, in normal host, I am not using any virtualbox, still i struck in loading the modules. Please do not top post. Install strace and then start asterisk with the command: # strace asterisk That should give you some low level info what's going on. More info about strace and available options can be found in: $ man strace Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2
On 27-08-12 08:25, Gopalakrishnan N wrote: This is really tuff working with OpenSuse. I am clueless how to sort our this. Maybe switch to a different distribution? I have used CentOS and RHEL for years without any problems and as far as I know both debian and ubuntu should work without problems too. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can we install 10 PCI card on asterisk
On 27-08-12 14:08, DHAVAL INDRODIYA wrote: Hi All, i would like to know if anyone has done or having idea regarding PRI terminations in asterisk. i have a requirement where i need to support 80 PRI in one machine i have found a machine which have 10 PCI slots available now i am thinking of arranging 8port sangoma card in this pci slots so i can arrenge 10 card in that. is it possible to run system like that ? is it good idea , can asterisk handle 2400 calls if machine size and RAM is good. I don't think Asterisk can handle that many DAHDI channels and I have never heard of an Asterisk box with more that 16 PRI's. Taking a step back, do you really want to put all your eggs in one basket? what if the box fails? That's 2400 channels going down and unavailable until you fix it. That will cost a log of money and get you angry clients. It makes more sense to spread the lot across different servers. Besides that, is your telco willing to provide you with 80x PRI or will they insist on aggregating it to several E3 links or something higher (STM-1)? If you really insist on going down this route have a look at FreeSWITCH or look at something like Cisco, Alcatel, Telco Systems etc. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding CHANNEL function values
On 25-08-12 14:31, Stefan at WPF wrote: Hello all, I need some help understand the values of the CHANNEL function, e.g. txploss // local packets loss rxploss // remote packets loss txjitter // local jitter rxjitter // remote jitter My main problem in understand is that a CHANNEL has two nodes (sender and receiver), while a typical setup includes at least 3 nodes: SIP phone - Asterisk - SIP Provider ( - each is a node) 1) So e.g. txploss, is it - what is lost between SIP phone and Asterisk - what is lost between Asterisk and SIP Provider - or probably both? I would assume that those statistics apply to a leg and not an end-to-end connection. So in your example I would assume that a txploss value is determined for the leg between the SIP phone and the Asterisk server and another txploss value is determined for the leg between the Asterisk server and the upstream SIP provider. Interesting stuff. If you figure it all out, please update this thread (and possibly the wiki). Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] xmpp / sip
Hi Hans, On 24-08-12 10:13, Hans Witvliet wrote: Hi all, After making a nice demo-setup for one of our innivationmanagers, he came up with a completely different stratagy ;-( Well if you could create it then obviously it's no longer innovative so they had to come up with something else :-) They want to have an Ejabberd server, with xmpp-clients. When you see a contact coming online, just point and click for making a phone call. The concept sounds like what Cisco was using internally. The Asterisk Wiki only mentions Tigase so you may want to verify that ejabberd supports the required XMPP PubSub stuff. https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+XMPP+PubSub Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 and 11
On 22-08-12 20:04, Giuseppe Longo wrote: Just a little questions, what's the difference between asterisk 1.8 and asterisk 11? Iirc you can check the ChangeLog in the Asterisk 11 sources. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2
On 14-08-12 08:29, Gopalakrishnan N wrote: If I change autoload=no then asterisk is starting, but post to that loading modules even chan_sip.so asterisk hangs. Its strange, only in OpenSuse I am facing this. In CentOS, Ubuntu its working fine, same Asterisk version with same hardware. Please do not top post and properly trim your replies. Have you made sure that on the OpenSuse box your DNS is configured properly? You should be able to lookup your IP address/FQDN both ways. So for example 192.168.1.1 (replace with your IP adres) should resolve in your.box.com (replace with your FQDN) and vice versa your.box.com should resolve into 192.168.1.1. See man dig or man nslookup for commands that can do DNS lookups. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Rackspace, My SIP phone behind NAT
On 10-08-12 10:12, SamyGo wrote: Oh, I see - check if your country blocks the SIP port 5060 ? try changing the default poert from 5060 to something else like and then try this. I think your ISP is blocking the SIP. If that is the case, setup an IAX connection and see if that works. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What TTS to use?
On 26-07-12 12:40, Chris Bagnall wrote: On 26/7/12 11:08 am, Ishfaq Malik wrote: I'm thinking about deploying TTS onto our asterisk servers and was just wondering which ones people use and like... We've tried Festival, Cepstral and Ivona. I didn't know about Ivona so thanks for mentioning it. Their UK English voice sounds very good. And with Cepstral's price hike with their release of version 6 (and killing of 5) it's good to know there's an alternative. Apparently Ivona has an GPL licensed Asterisk plug-in (http://www.ivona.com/en/telecom/asterisk/). Do you know where that plug-in can be downloaded? I could not find it. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] freepbx asterisk
On 20-07-12 09:15, neo nortan wrote: dear i am a neewbi for asterisk, plz tell me or if any link is there where i can understand how asterisk, freepbx, web-meetme, dahdi all these tools works and how they are related. plz help me. http://www.asteriskdocs.org/ Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?
Hi, The flowroute website mentions that they set callerid on outbound calls based on the presence of (in order of preference): P-Asserted-Identity, Remote-Party-ID or From:. I've been trying to make outbound callerid work via flowroute to no avail. Does anyone have an extensions.conf / sip.conf snippet howto make this work? This is for Asterisk 1.4.44. Thanks! Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?
On 10-07-12 18:29, Alex Balashov wrote: SIPAddHeader() comes to mind. :-) Yup I got that far :) I tried things like (with correct name number): exten = _1ZX,1,SipAddHeader(P-Asserted-Identity: Global Minties Corp sip:19995551212@AST_BOX_FQDN) But that did not work as flowroute always sends VOIP CALLER and a ton of different numbers on outbound calls. So I guess I am doing something wrong but I can't figure out what. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?
On 10-07-12 18:49, Warren Selby wrote: You can't* set the outbound name. That's defined in the national caller id name database that the receiving phone company dips into. As far as I know, Flowroute does not add entries to this database, nor do they dip it when you receive a call to pass the caller ID name on inbound calls. Other providers do. Thank you for your feedback Warren. I removed the outbound name but still get random numbers VOIP CALLER on outbound calls. Googling I tried some more: SipAddHeader(P-Asserted-Identity: sip:19995551212@AST_BOX_FQDN) SipAddHeader(P-Asserted-Identity: 19995551212) SipAddHeader(P-Preferred-Identity: sip:19995551212@AST_BOX_FQDN) SipAddHeader(P-Preferred-Identity: 19995551212) But none of them work. So unless someone has the magic incantation howto make this work I'll open a ticket with flowroute. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?
On 10-07-12 18:48, Danny Nicholas wrote: Check your users.conf - this looks like an override issue to me. Thank you for your feedback Danny. users.conf is default and has not been touched. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?
On 10-07-12 18:47, Ira wrote: At 09:20 AM 7/10/2012, you wrote: I've been trying to make outbound callerid work via flowroute to no avail. Does anyone have an extensions.conf / sip.conf snippet howto make this work? This is for Asterisk 1.4.44. This is a section of code I use to choose outgoing callerid for my Flowroute lines. I have a number of companies and this lets the caller select what the called parts sees. Ira same = n(got0),set(thiscid=NOONE2345678901) same = n,goto(gotcallerid) same = n(got1),set(thiscid=Bob and Lucy3124726322) same = n,goto(gotcallerid) same = n(got2),set(thiscid=31247240223124724022) same = n,goto(gotcallerid) same = n(got3),set(thiscid=Mustang3126925021) same = n,goto(gotcallerid) Thank you for that snippet Ira. You and Warren were spot on. All is fine now. Thanks! Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flowroute: howto set outbound callerid (ast 1.4)?
On 10-07-12 19:48, Warren Selby wrote:
On Tue, Jul 10, 2012 at 12:34 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl
mailto:asterisk-l...@puzzled.xs4all.nl wrote:
Thank you for your feedback Warren. I removed the outbound name but
still get random numbers VOIP CALLER on outbound calls. Googling
I tried some more:
SipAddHeader(P-Asserted-__Identity: sip:19995551212@AST_BOX_FQDN__)
SipAddHeader(P-Asserted-__Identity: 19995551212)
SipAddHeader(P-Preferred-__Identity: sip:19995551212@AST_BOX_FQDN__)
SipAddHeader(P-Preferred-__Identity: 19995551212)
But none of them work. So unless someone has the magic incantation
howto make this work I'll open a ticket with flowroute.
I use Flowroute. My outbound callerID is set as follows:
[outgoing]
exten = _X.,1,Verbose(Outound call from ${callidnum} to ${EXTEN} on
${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)})
exten = _X.,n,Set(CALLERID(num)=${callidnum})
exten = _X.,n,Goto(outgoing-dial,${EXTEN},1)
[outgoing-dial]
exten = _NXXNXX,1,Dial(SIP/1${EXTEN}@flowroute)
exten = _1NXXNXX,1,Dial(SIP/${EXTEN}@flowroute)
${callidnum} is a variable from my SIP peer
(setvar=callidnum=7133437300). This always passes my proper phone
number when I make outbound calls.
Thank you for that snippet Warren. I setup a different US DID and called
that one via flowroute and the callerid worked. Previously I called a
voip.ms toll-free number. I'll blame it on (lack of) carrier
interoperability :) Good to know outbound callerid works without having
to use magic SipAddHeader incantations.
Thanks!
Patrick
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Re: [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash
On 10-07-12 20:42, Carlos Alvarez wrote: I'm currently trying to decide on which GUI-enabled version of Asterisk to use for one particular installation, where we will need good telecommuter support. We've made it so easy for people to work remotely that the customer is downsizing their real estate and will have 90% remote workers with them rotating through the office as needed. So most phones in the office will be shared, and I'm looking for a version of Asterisk that will easily allow people to log in and out of a specific desk. What are your suggestions? I have very little experience with GUI versions of Asterisk; we use bare Asterisk for nearly everything. Afaik development of the free Trixbox has stalled. Elastix is a popular alternative and iirc it comes with provisioning tools that work with Polycom. Or you could just install a bare Asterisk and slap FreePBX on top of it if you don't need all the fancy stuff that Elastix offers. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMR - Segmentation Fault
On 09-07-12 12:00, Chandrakant Solanki wrote: Hi @Patrick, are you using which AMR source, will you please provide me link, I also tried with 1.8.11 but didn't found success. I am using sourceforge one http://sourceforge.net/projects/aterisk-amr/files/ I used the one from sourceforge: http://sourceforge.net/projects/aterisk-amr/files/ I can't help you any further since I deleted everything after I noticed that gsm and ilbc codecs were pretty similar to amr, at least in my situation. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Centos 6 mISDN
On 09-07-12 07:42, Andrew Colin wrote: Hi Patrick Can you possibly guide me as to how you got it working. I am running the same version of Centos and asterisk. Did you use a specific kernel? I did not use a specific kernel version. I got the latest versions from misdn.eu and installed those. Once you have done that have a look at the doc for LCR at http://isdn.eversberg.eu/ which is a bit dated but may help in setting up LCR. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Centos 6 mISDN
On 03-07-12 09:38, Andrew Colin wrote: Hi Guys Has anyone got this working on Centos 6? I set it up on a CentOS 6.2 x86_64 box with Asterisk 1.8.11 and the latest mISDN/mISDNuser/isdn4k-utils/lcr stuff and made some test calls between a Polycom 650 and a GSM calling into the ISDN line attached to a HFC-S based TA. Seemed to work fine although I only did very limited testing. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Another IP address to block
On 06-06-12 11:41, Thorsten Göllner wrote: Where can I find such ip-lists, please? http://www.ipdeny.com/ Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for some quality testers for zoiper softphone for android.
Hi zoa, On 31-05-12 17:39, joachim wrote: Ellow, We released zoiper for Android today, available for free here: https://play.google.com/store/apps/details?id=com.zoiper.android.app SIP and IAX is supported, should work quite well, unfortunately it is really hard to test all android and hardware combinations. Any android lovers out there to send us some feedback ? Preferably with packet capture skills ? I am mainly looking for feedback on the audio quality, audio delay and if everything looks ok in the gui. Had a quick look on Google Play. Are g729, AMR-NB and SRTP missing in the description or does it not have those features (yet)? Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] British Telecom ISDN BRI line issues
Hi Khalid, On 18-05-12 20:50, khalid touati wrote: Hi Patrick, it seems like you have the magic ball, I think what you described is exactly what happened: After we tested the server+ link and we were able to have simultaneous calls (as expected), and knowing that this server was not touched (not even rebooted), it is back not dialing through PTP link.The server remained working from Friday to at least Monday then boom, when I called BT...of course no explanation I am just wondering: their mechanism to miss up things, is it automatic or manual ( I think automatic). As far as I know the telco's monitoring infra will automagically disable the ISDN port. You can request that the telco disable monitoring of the port which will prevent them from shutting it down. But if there's something wrong with the port/linecard on their switch they will no longer notice. In the end the only solution is to deploy software than can properly handle the D-channel in all its states. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set iaxmodem receiving speed
On 16-05-12 17:10, gincantalupo wrote: Hi Larry, thank you for your answer. This is same test I did. After this I lowered again to 4800...result: iaxmodem receives at 9600 b/s.that's why I cannot solve the puzzle. I put that line (Class1RMQueryCmd: !24,48,72) in config.IAXtty and tried other configuration files too but I always get faxes at 9600. Did you restart iaxmodem and hylafax after you made those changes? Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] British Telecom ISDN BRI line issues
On 05/13/2012 03:07 AM, khalid touati wrote: My Issie is finally fixed and I can make calls, I received actually from digium the fix, I'll try to give as much details as I can to make sure people who find this thread understand pb and solution. Problem: not able to dial calls using BRI from British Telecom configured as system access or PTP (standard access which is PTMP works fine) (see above for errors thrown) Solution: adding this line options wctdm24xxp bri_persistentlayer1=1 to /etc/modprobe.d/dahdi.conf and restarting asterisk and dahdi of course. Versions used: asterisk 1.6.2.6 dahdi 2.3.0.1 libpri 1.4.10.2 Good to hear you got it working with Digium's help. One thing: if bri_presistentlayer means that the drivers will force the D-channel to always be up then do not be surprised if BT disables the ISDN port. They don't like it if a customer forces them to power the D-channel all the time at their expense. And with their ISDN team not contactable it may be a bit of a challenge to get them to enable the port again (after you promised not to mess with heir D-channel again...). Special thanks to Patrick! thanks and good luck to all! My pleasure. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] British Telecom ISDN BRI line issues
On 11-05-12 05:44, khalid touati wrote: Thank you for your reply Patrick! for the first situation, I did try asterisk 1.6.2.6 and dahdi 2.3 but with no success. Can anyone suggest a combination that works till a patch is released? Unfortunately not and I don't have a Digium BRI card to test with anyway. Perhaps you could ask Kevin for the patches so you can participate in the testing and help making sure it works properly? Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] British Telecom ISDN BRI line issues
On 10-05-12 21:10, khalid touati wrote: Hi All, I did downgrade to asterisk 1.6.2.6 and dahdi 2.3 which is working with other BRI line (in our NL office), but I get this type of errores: -- Called G1/0788744550 [May 10 20:07:20] ERROR[27479]: chan_dahdi.c:12280 dahdi_pri_error: ACK received for '1' outside of window of '0' to '0', restarting == Primary D-Channel on span 3 down [May 10 20:07:20] WARNING[27479]: chan_dahdi.c:4158 pri_find_dchan: No D-channels available! Using Primary channel 9 as D-channel anyway! == Primary D-Channel on span 3 up [May 10 20:07:20] ERROR[27480]: chan_dahdi.c:12280 dahdi_pri_error: Huh!? no master found I didn't change my config in my previous post, anyone familiar with this type of errors? No but there is a bug report with a lot of information that seems similar: https://issues.asterisk.org/jira/browse/14031 In Europe telco's drop the D-channel (cut off power) to save on the electric bill. The libpri/dahdi/asterisk combo should detect a dropped D-channel and signal the telco to fire up the D-channel. Judging from that bugreport (Unresolved) it seems Digium has still not succeeded in properly handling this situation. Should you not be able to resolve this issue and really require an ISDN BRI connection then have a look at an Eicon Diva or Sangoma card. Both cards+drivers properly handle a dropped D-channel. I have used Eicon Diva and Sangoma BRI cards in the Netherlands and they work fine. Or you could get a blackbox from Beronet, Patton, Audiocodes etc. If you feel adventurous you can also get a BRI card with a HFC-S Cologne chipset and get the latest and greatest isdn4k-utils, mISDN, mISDNuser and lcr from http://misdn.eu, build, install and configure lcr to talk to asterisk. A few weeks ago I set it up and did one test call and that call worked fine. Use at own risk :) Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
