Re: [asterisk-users] Asterisk with PABX
I'm in the hybrid group. I looked at TrixBox for a couple of weeks and figured out how to add my ATA and got a VoIP DID working, all very neat.Then I installed the svn trunk version of *. Then I proceeded to install FreePBX on top of my * install. The reason I did it was because I was using VoicePulse Connect and the TrixBox has a nice little module for it. I actually got the module to work on my */FreePBX hybrid, but have since had to just add the trunks and outbound routes manually due to some weird occurences (some incoming calls not authenitcating properly, etc.) I still use FreePBX for this, but am definitely learning a LOT in the conf files. Regards,Sugar Dave FreemanOn 8/28/06, Dean Collins [EMAIL PROTECTED] wrote: Hey if you don't need the work just say no, I'm sure someone else willbe happy to take the money from them. Maybe the monkey?Cheers,Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Rushowr Sent: Monday, 28 August 2006 2:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk with PABX Too true too true Personally, I think trying to use Trixbox tolearn Asterisk is akin to a monkey humpin' a footballIt's just not right. Anywhohad to do my smartass deed for the day Rushowr (Hates getting contracts to fix someone's AAH/TrixBox/FreePBX phone system) -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Eric ManxPower Wieling Sent: Monday, August 28, 2006 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk with PABX Dean Collins wrote: Yes it is possible. May I suggest you spend more time with www.voip-info.org Or even better download www.trixbox.org on an old server to get an idea of how configs work. Getting Trixbox would help him understand how Trixbox configs work, not how Asterisk configs work. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LOUD MP3 Hold Music
I have the opposite problem. I can hardly hear the hold music at all.On 8/22/06, Dennis P. Clark [EMAIL PROTECTED] wrote:How do you lower the volume of MP3 hold music?Dennis Clark DENPROWRK 207.618.1998CEL 443.415.0527FAX 1.888.811.8809[EMAIL PROTECTED]___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
You might be able to use virtual NICs to eliminate the problem with non-standard ports for a company's SIP phones. Or real NICs using a couple of multi-homed cards.I haven't tried it, though. On 8/16/06, Douglas Garstang [EMAIL PROTECTED] wrote: Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc? Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance.Doug.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
Then virtual would be the way to go...I'm no expert, so you'd have to do some research on how many virtual interfaces you could use reliably.But some of the other suggestions I've seen might be a better option? Separate contexts for each entity, etc. On 8/16/06, Douglas Garstang [EMAIL PROTECTED] wrote: Well, we're talking about several dozen, maybe 100, companies, per Asterisk box here. -Original Message-From: David Freeman [mailto:[EMAIL PROTECTED]]Sent: Wednesday, August 16, 2006 11:36 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Asterisk 'Hosting'You might be able to use virtual NICs to eliminate the problem with non-standard ports for a company's SIP phones. Or real NICs using a couple of multi-homed cards.I haven't tried it, though. On 8/16/06, Douglas Garstang [EMAIL PROTECTED] wrote: Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc?Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance.Doug.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with incoming authentication
I'm far more interested in finding out a solution to my problem, but:1 When everything is set up correctly, they're just fine.2 Yes, except for the problem I originally posted to start this thread.3 I don't know, it all depends on you. 4 I don't know, VoicePulse is the only think I've tested with so far.On 8/15/06, Crazy Boy [EMAIL PROTECTED] wrote:Hi,I am Chandra from India. We have Installed Asterisk in our organization. We want to buy a VoIP plan to make calls to US. I have some doubts. Please clarify. 1) How is the Voicepulse service?2) Is Voicepulse working fine with Asterisk?3) Can I configure Voicepulse easily with Asterisk?4) From Teliax and Voicepulse, Which is offering better service?Looking forward to your response. Thank you. Regards,Chandra.David Freeman [EMAIL PROTECTED] wrote: Sometimes I can receive a call to my DID, but sometimes it just rings and rings and I see these messages in the full log:[Aug 14 23:32:12] DEBUG[3556] res_crypto.c: Key failed verification: voicepulse20060419 [Aug 14 23:32:12] NOTICE[3556] chan_iax2.c: Host 64.61.93.87 failed to authenticate as voicepulse[Aug 14 23:32:13] DEBUG[3550] res_crypto.c: Key failed verification: voicepulse20060419[Aug 14 23:32:13] NOTICE[3550] chan_iax2.c: Host 64.61.93.90 failed to authenticate as voicepulseUsually, when this happens, I can immediately re-dial the DID and * receives the call and sends it on the dialplan.I've tried configuring my IAX2 user details to no use rsa and no key, but the problem persists. I've tried to delete my IAX2 trunks to just use the SIP ones and I get the same problem...in fact, if I only have SIP trunks, I can't receive any calls to the DID.I'm using Asterisk SVN-trunk-r39753M currently, updated today. Any help would be appreciated, I'm running out of ideas. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Stay in the know. Pulse on the new Yahoo.com. Check it out. Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with incoming authentication
Sometimes I can receive a call to my DID, but sometimes it just rings and rings and I see these messages in the full log:[Aug 14 23:32:12] DEBUG[3556] res_crypto.c: Key failed verification: voicepulse20060419[Aug 14 23:32:12] NOTICE[3556] chan_iax2.c: Host 64.61.93.87 failed to authenticate as voicepulse[Aug 14 23:32:13] DEBUG[3550] res_crypto.c: Key failed verification: voicepulse20060419[Aug 14 23:32:13] NOTICE[3550] chan_iax2.c: Host 64.61.93.90 failed to authenticate as voicepulseUsually, when this happens, I can immediately re-dial the DID and * receives the call and sends it on the dialplan.I've tried configuring my IAX2 user details to no use rsa and no key, but the problem persists. I've tried to delete my IAX2 trunks to just use the SIP ones and I get the same problem...in fact, if I only have SIP trunks, I can't receive any calls to the DID.I'm using Asterisk SVN-trunk-r39753M currently, updated today. Any help would be appreciated, I'm running out of ideas. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] safe_asterisk to start latest version from SVN - trying asterisk with googletalk
I had a similar error, and another on format_au (that one just never compiled) so I had to noload the au.On 8/11/06, Hadley Rich [EMAIL PROTECTED] wrote:On Saturday 12 August 2006 14:30, Marco Mouta wrote: [Aug 12 03:27:35] VERBOSE[26610] logger.c:[format_mp3.so][Aug 12 03:27:35] WARNING[26610] loader.c: missing mod_data for format_mp3.so What could be wrong?Looks like you have an old format_mp3 module in your module directory. Removing this (and any other old modules) or adding a noload to modules.confshould fix the problem.If you compiled trunk it should have given you a big fat warning about oldmodules on installation. hads--http://nicegear.co.nzNew Zealand's VoIP supplier___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Autoreply: [asterisk-users] Tri-Link Technologies?
It's a horrible, horrible autonotice that this person is unavailable. Expect to see lots of these.To contribute to the topic, I also can't find much on this phone ;)Dave On 8/9/06, Don [EMAIL PROTECTED] wrote: Dude I am English- Original Message -From: [EMAIL PROTECTED]To: asterisk-users@lists.digium.com Sent: Wednesday, August 09, 2006 12:56 PMSubject: Autoreply: [asterisk-users] Tri-Link Technologies?Attualmente non sono in sede. Perrichieste urgenti contattare lo 800919299 o inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED].Cordiali SalutiGiuseppe ParlatoArea Networkmailto:[EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users--No virus found in this incoming message.Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.10.8/414 - Release Date: 8/9/2006___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Probelm with IAX peers
So, I've been testing with VoicePulse Connect for awhile, and am getting more confident in setting up dialplans, extensions, etc.Now, last night I signed up for an account with Vitelity. I don't have a DID yet, as it takes a million years to get your custom toll-free DID, but I digress. Following the instructions (very limited) that are provided by Vitelity I have 2 IAX friend connections.My problem is this: After a short amount of time, usually one call to my VoicePulse DID, ALL calls to the VoicePulse DID stop coming in to *. If I nix the Vitelity friends everything works again. I have created the Vitelity contexts in iax_custom.conf (I'm using FreePBX to manage my extensions, etc.) and think this is the right way to go.Is there any reason why * would stop listening on my first two IAX peers in favor of only listening to IAX friends? If anyone else is using Vitelity (specifically with FreePBX management) is it NECESSARY to use the vitelity-inbound and vitelity-outbound contexts? I'd be interested in seeing your configuration.Thanks, Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: * and GTalk testing
Got blasted for sending this to the wrong list, anybody using GTalk extensions?-- Forwarded message --From: David Freeman [EMAIL PROTECTED]Date: Aug 7, 2006 10:11 AMSubject: * and GTalk testingTo: asterisk-dev@lists.digium.comOkay, I'm a complete novice at *, but I'm learning. I started with a TrixBox demonstration and have since moved on to an SVN build of *. So far, I've been able to get FreePBX installed so I can have a quick way to configure my VoicePulse Connect account and have a quick way to add extensions, etc. I decided to see if I could get * to call/receive GTalk clients. I'm almost there. I can successfully contact a GTalk client AND I can successfully get a GTalk client to call an extension in *. However, the problem I have is that there is no audio coming through either direction. A quick test of GTalk - GTalk shows that audio is coming through, so I'm looking for pointers to help track down what's going on in *.Thanks,Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: * and GTalk testing
I don't think this is realted, but I had to eliminate zttranscode from being compiled as it kept killing * after a few seconds. Since I'm not using any pstn (or hardware for tha tmatter, ztdummy) I don't think I need it? Like I said, still learning here, but I want to make sure I cover all the oddities I had to do. Also, there was no format_au.so after building the SVN trunk? Is this normal?On 8/7/06, Matt Riddell (NZ) [EMAIL PROTECTED] wrote:-BEGIN PGP SIGNED MESSAGE-Hash: SHA1 David Freeman wrote: Got blasted for sending this to the wrong list, anybody using GTalk extensions?:)Yes and no,I've written a bot that responds to messages, but have thesame problem as you with audio. - --Cheers,Matt Riddell___http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community)http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)-BEGIN PGP SIGNATURE-Version: GnuPG v1.4.2 (MingW32)Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.orgiD8DBQFE122FS6d5vy0jeVcRAmaXAJ419kfuhKwgsmBIQdx9TcNI04rfqwCfVD6tZ+iVW73SSMuxnvjAhWagY5c= =oLWU-END PGP SIGNATURE-___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trunk transferring?
Hello all.I'm very new to the list and quite new to Asterisk.I did a little messing around with a prebuilt system (TrixBox) while investigating phone systems for a business that my partner and I are getting together. Anyway, what I'm wondering is if it's possible to do this:A call comes in to a US 800 number that is a SIP or IAX VoIP line, then * can transfer that incoming call to use another SIP or AIX VoIP line? I want to offer an 800 number (and pay for the initial time for the call) but I don't want to pay 800 charges for the duration of the call (we could be on the phone with someone an hour or longer.)I've seen references to trunk transferring in some commercial products for *, but I'm thinking if they have that functionality, I should be able to build it, too! Thanks in advance.Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trunk transferring?
Oh yeah, I forgot to say that I get incoming calls for free on the trunk I want to transfer the 800 call to.You do make a good point for how to keep costs lower if not as close to free as I would like. On 7/27/06, Manrique Feoli [EMAIL PROTECTED] wrote: why not calling them back?1- you get the call on the 800 line, and after the operator evaluaesthe situation,dials a digit or something, the system calls back thesame number,but via your preferred route/system/billing. 2- get the call, then play an automated message to the user explaininghow and why we'll call him back in a minute,so he hangs up and waitfor the callDavid Freeman escribió: Hello all. I'm very new to the list and quite new to Asterisk. I did a little messing around with a prebuilt system (TrixBox) while investigating phone systems for a business that my partner and I are getting together. Anyway, what I'm wondering is if it's possible to do this: A call comes in to a US 800 number that is a SIP or IAX VoIP line, then * can transfer that incoming call to use another SIP or AIX VoIP line? I want to offer an 800 number (and pay for the initial time for the call) but I don't want to pay 800 charges for the duration of the call (we could be on the phone with someone an hour or longer.) I've seen references to trunk transferring in some commercial products for *, but I'm thinking if they have that functionality, I should be able to build it, too! Thanks in advance. Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dlink DVG 1120S/Asterisk VoIP to PSTN
Okay, I've looked for some days and only ever find other people asking this question and no responses (I did see a few e-mail me answers, but those haven't panned out for me).Is it possible to use a D-link DVG 1120M (flashed to SIP firmware version v 0.0-S08) to make PSTN calls from Asterisk?I can use the device with two phones as SIP extensions, and I have it configured to allow the phones attached to the ATA to dial # and get a dialtone on the PSTN.It seems I am close, but something is not clicking with me. Any help is appreciated.Thanks,Sugar Dave Freeman ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
