[asterisk-users] Please unsubscribe or moderate [EMAIL PROTECTED]
All these repeated list replies with Autoreply: Autoreply: Autoreply: Autoreply:... subjects are irritating at best and debilitating at worst! This makes the list waste bandwidth and my inbox (and the archives too) unreadable! Thx! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN^H^H^H^H^HmISDN 2 Sweex HFC-PCI cards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The downside of Asterisk and least cost routing...
On Fri, May 11, 2007 08:21, Gordon Henderson wrote: On Thu, 10 May 2007, Francesco Peeters (Asterisk) wrote: If you think your ISP is reliable enough then go for it! I've had less ADSL issues last year than ISDN issues! ;-) (And that while ADSL is running over that very ISDN line!) There is a small (and growing!) number of small businesses (and not so small ones either!) who are moving towards using their broadband (typically ADSL in the UK) connection for Telephony - and even installing a 2nd ADSL line just for VoIP. It can work out a lot cheaper than going down the traditional ISDN2/ISDN30 route for a lot of people as a small business expands. I can see that would work out that way, yes! Undfortunately I'll have to pay reconnection fee before I can cancel! :-o I guess that's a country thing - good luck :) I found out that I can even transfer my current main number to my ISP's SIP service for EUR 5 a month... Aside from that they can give me 2 free incoming numbers in the 087 range, and I already have an incoming VoipBuster number in my own areacode... That would give me 4 incoming numbers... The only thing I'd probably lose is the ability to do faxes! So I am going to investigate that further first! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] The downside of Asterisk and least cost routing...
On Fri, May 11, 2007 10:31, Chris Bagnall wrote: There is a small (and growing!) number of small businesses (and not so small ones either!) who are moving towards using their broadband (typically ADSL in the UK) connection for Telephony - and even installing a 2nd ADSL line just for VoIP. Indeed, many of our clients are doing just that. I would, however, strongly recommend against ditching PSTN entirely (in the UK, it's virtually impossible anyway since ADSL requires a PSTN line over which to run) - those PSTN lines are still useful for things like emergency service calls, directory enquiries, etc. etc. In NL you actually can ditch the telephony and keep the ADSL... My ISP even gives emergency access if you transfer your main number to their SIP service. And there still is my cell-phone too! ;-) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] The downside of Asterisk and least cost routing...
I forgot to pay this month's phone bill, and never noticed until family (the in-laws, who are too cheap to try the cell phone if landline fails, because it is 'more expensive') told me they were unable to reach us... As it turns out, the phone company disconnected us, but because Asterisk routes all outgoing calls in the Netherlands over VoipBuster, I never noticed anything! ;-) If I'd given out my VoipBuster DID, I'd probably still wouldn't know! *ROFLOL* It gives me pause though... Maybe it's time to get rid of my fixed line... ;-) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] The downside of Asterisk and least cost routing...
On Thu, May 10, 2007 23:44, Gordon Henderson wrote: On Thu, 10 May 2007, Francesco Peeters (Asterisk) wrote: It gives me pause though... Maybe it's time to get rid of my fixed line... ;-) No ;-) needed - I have friends on cable internet with no separate copper phone line now. I'd consider it myself if I weren't tied to having ADSL over my phone line, and as yet there isn't a way to separate them (in the UK) In NL there is... ;-) Especially interesting as I have ISDN, which is almost twice as expensive... So I am really going to look in to it... I'd save about EUR 20,00 per month that way! Undfortunately I'll have to pay reconnection fee before I can cancel! :-o -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] Any other softPBX like Asterisk?
On Fri, May 11, 2007 07:34, Armin Schindler wrote: On Thu, 10 May 2007, Crazy Boy wrote: Hi Friends, Can anybody tell me other softPBX softwares like Asterisk? - OpenPBX - Freeswitch Or try Googling for something like 'open source pbx'... Sheesh! :-o -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] freepbx - DB Error messages...
On Sat, March 24, 2007 19:10, Bruce Reeves wrote: You might get a faster response on freepbx/amp mailing list. On 3/24/07, Francesco Peeters (Asterisk) [EMAIL PROTECTED] wrote: SNIP Just an update: Still have NOT been approved for either the mailing list *or* the forum! I am pretty disappointed in the moderators! If you take up the responsibility to moderate a list or forum you have to make sure you respond promptly, especially if the list or forum (or both) require moderator approval before a user-account is activated! (And no, my original answer has not been answered yet either!) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 VIA EPIA V8000 - 256 MB - * 1.2.4 - mISDN, but still no freePBX 2 Sweex HFC-PCI cards ___ --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] error in FreePBX
On Thu, March 29, 2007 19:36, Carlos Jerónimo wrote: Hi Steve, your sugestion is correct, but i registed 2 times in FreePbx foruns this week, and my login is inactive yet. In the mail i receive this msg: Welcome to FreePBX Forums Forums Please keep this email for your records. Your account information is as follows: Your account is currently inactive, the administrator of the board will need to activate it before you can log in. You will receive another email when this has occured. Same here... Been waiting a week since my last attempt, but still nothing!... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] freepbx - DB Error messages...
Hi all, I am probably missing something ultimately obvious, but I have a problem configuring freepbx... Using Edgy Eft with the cvs freePBX 2.2.1 and followed the Ubuntu installation guide on freepbx.org. System pxe-boots from a server with NFS root on same Using * 1.2 current (from source, not .deb's) Using mISDN-streams (from source, not .deb's) Using freePBX-2.2.1 (from source, not .deb's) Installed everything, and mISDN and * load just fine amportal start works fine as well However I keep getting DB Error's in the GUI... The syslog gives two separate errors: 1) Error 127 when reading table ./asterisk/whatever 2) Table is crashed and needs to be repaired I created a special mysql user for * and did an PERMIT ALL PRIVILEGES on the mysql databases When I log in to mysql as root and do 'SELECT username FROM ampusers ORDER BY username' I get the record list. When I do the same as the * user, I get the 'Table is crashed, blablabla' line. I tried changing the login user for freepbx (ampdbuser) to root, but that doesn't help either, as I keep getting the 127 error... Googling wasn't very helpful, and the freepbx forum admins still haven't approved my account, so I thought I'd try here... Any help appreciated! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] Issue with Hamlet ISDN PCI card(Cologne Chipset)
On Sat, March 24, 2007 11:54, Mauro Zanin wrote: Hi everybody I have installed a TrixBox with Asterisk 1.2.14 and relative upgreaded software. I Bristuffed it with last version of bristuff to use a Hemlet PCI ISDN CARD in a normal Italian EUROISDN installation. The * works fine except for the ISDN CARD. It is always Channel D down, but if a Call comes in, it works perfectly for some time, both inbound and outbound. It prompts Channel D UP! If I disconnect the NT+ termination the Channel D goes down at once. Did I make something wrong? Not really... It's a bristuff quirk... It doesn't gracefully handle the forced D-channel down that most European ISDN operators implement. That is why I switched to testing vISDN, but that has been stagnant for over half a year without any fixes for a few very annoying bugs, because the programmer dedicated all his time to rewriting the vGSM part... I am now testing mISDN as someone on the vISDN list mentioned that it's chan_misdn voice support had greatly improved... The only way I can *somewhat* keep bristuff working without contacting the ISDN carrier to turn on the D channel permanently is by initiation a 100ms outbound call every minute using the manager interface... (Yes, a very ugly kludge indeed, but I do not want permanent channel up, as I want to be able to test everything in a normal environment, as I am planning to install this in other location too once I have a stable, reliable environment) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] IAX softphones
On Wed, October 18, 2006 19:03, Paul Gaffney wrote: Hi, can anyone recommend a good IAX phone for use with Asterisk? I'm looking for a NAT-friendly solution and my SIP phones are good but not dependable. Neil Neil, www.asteriskguru.com http://www.asteriskguru.com/ lists a few of them. Try IDEFISK. Paul Gaffney LANStatus,LLC I personally like DIAX on for Windows users. Haven't yet found an IAX phone I like on Linux... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] IAX softphones
On Wed, October 18, 2006 21:07, Guillermo Salas M. wrote: On Wed, 2006-10-18 at 20:08 +0200, Francesco Peeters (Asterisk) wrote: On Wed, October 18, 2006 19:03, Paul Gaffney wrote: Hi, can anyone recommend a good IAX phone for use with Asterisk? I'm looking for a NAT-friendly solution and my SIP phones are good but not dependable. Neil Neil, www.asteriskguru.com http://www.asteriskguru.com/ lists a few of them. Try IDEFISK. Paul Gaffney LANStatus,LLC I personally like DIAX on for Windows users. Haven't yet found an IAX phone I like on Linux... Kiax works great with Gnome, KDE or Xfce. -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 I'll try that later, thanks! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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 behind Sonicwall firewall
On Tue, September 26, 2006 22:21, Barry Fawthrop wrote: Hi all I didn't change anything that's my point It has be running and working just fine then at 4:32 pm yesterday I could not make or recieve VoIP calls via our VoIP Provider They say the Invite packet was being rejected and thus there was no real connection even though SIP SHOW PEERS has us registered They also say it's due to the Sonicwall which has changed port assignments and thus blocking ports. I see in the Sonicwall log UDP Packet Dropped with the Providers IP Address but it talks about port 36612 which is not SIP They say along with the log that SIP is using 36612 why when all the VoIP SIP setting are enabled/configured and SIP is packet forwarded to the Asterisk Box due to Sonicwall NAT Now I'm trying to find out why and how to correct this. Thanks all Barry SonicWALL Enhanced has an option called 'Persistent NAT'... Is it turned on? -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] Any Hardphone with VPNClient embedded?
On Mon, September 4, 2006 16:55, Cory Andrews said: Please be aware that from a future support standpoint, you may be a bit limited with Zultys. Their future seems very uncertain they have recently just about ceased operations and let the majority of their employees go. Cory J Andrews voice - 800.398.VoIP X3402 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: Leo Ann Boon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, September 04, 2006 10:35 AM Subject: Re: [asterisk-users] Any Hardphone with VPNClient embedded? Marco Mouta wrote: Hi all, Does any of you knows an Hardphone with VPN client embedded? Take a look at Zultys SIP phones. VPN enabled. www.zultys.com As I too am interested in IPsec capable hardphones (or ATA's), do you have a suggestion what to look at instead? I mean: It's nice to say the company may not be around for long, but if there's no alternative, what choice does one have? TIA! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] VoipNow 1.2.0 Beta
On Mon, July 31, 2006 21:44, Tom said: At 02:21 PM 7/31/2006, you wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Senad Jordanovic wrote: [EMAIL PROTECTED] wrote: Tom Vile wrote: Did you look on the site? http://www.4psa.com/products/voipnow/demo.php Does above means that the license for voipnow need to be paid to packet 8 as well? http://biz.yahoo.com/prnews/060613/sftu062.html Senad Hate replying on my post but what a heck!!! My understanding is that ANY hosted IP PBX coded in any object oriented programming language is falling under the above mentioned patent. Anyone has any thoughts on this? Another reason not to do business in the USA! Any good suggestions on where to buy rack space in a country that is not honoring stupid US patent law and has great and secure Internet connections? Tom Ehrm... Russia, China... You could also try several European countries, such as the Netherlands, Luxembourg, Switzerland... I just have mine at home... Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On Wed, June 28, 2006 10:14, [EMAIL PROTECTED] said: Well, look at it this way: if you get the working, you can buy one of those tiny form-factor 386 boards with the 2 pcmcia slots and get a pcmcia soundcard and a ethernet port. Run Linux off a CF card and have it setup to *only* interface with Skype and Asterisk. Basically, make a Skype ATA, but it would convert Skype to SIP. I think that could still be considered an ATA, right? Or a gateway at least. Since you can make a Skype account for free and can (for right now) make US and Canada LD calls for free, I think the cost and time to make them would be worth it. :) And if you figure out a good price for them, people might even buy them from you Undrhil Another advantage is that you can reach all those people who have Skype and are not willing to try Voipbuster or similar SIP based providers, and tell them that SIP/IAX/Asterisk *is* the better solution, because they cannot do the same with Skype the other way round! ;-p -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] Oh oh. Micro$oft just noticed VoIP
On Tue, June 27, 2006 0:26, shadowym said: They have been talking about this for awhile. If you look at the real time and embedded operating system world they have not really done so well over the many years they have been trying. Just throwing money at the problem has never worked for them in the past either. Perhaps because people expect devices like that to Just Work(tm), something Embedded Linux is better known for than Embedded Windows is?... The Asterisk community has nothing to worry about in the near term if ever IMHO. Unless they buy Digium... That'd give them a serious amount of code to obfuscate and hide in closed source products! ;-) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On Mon, June 26, 2006 20:06, Brian Capouch said: Tzafrir Cohen wrote: On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? É freeware? Podemos seguir com o projeto Asterisk-PT? English, please, folks. Let them talk. What's it hurt the rest of us? It is more a question of netiquette... If you're on an English mailinglist, you should speak English (Not attacking Josué and Marco, just answering the question here). It is not only more productive (If you keep to English, more people understand and can contribute to *and* profit from the discussion), but speaking a different language not spoken by the majority on list is generally considered akin whispering in company: not quite rude, but also not-done... We have seen the wages of tortured English sometimes unleashed on the list. If they're getting the job done, I say hit the Delete button and get on with your life. You can hit the delete button for bad English too, you know! ;-) If 80% of the list traffic were in foreign languages, then I would say we would have an issue. Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten! ;-) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] David Choo/eServices/eSpore is overseas
On Mon, June 12, 2006 4:37, David Choo said: I will be out of the office starting 12/06/2006 and will not return until 17/06/2006. Dear Sir / Mdm, I'm currently travelling. During this period of time, I have minimal access to internet and email. As such, please be aware that I might not be able to reply to your queries promptly. I apologise for the inconvenience caused. SNIP Tongue mode='in cheek' That is good to know! We will start monitoring your residence until we find an opportune moment to enter. We will then lend a hand in (re)moving the most precious of your things to a new address... /Tongue (Sorry, couldn't help myself!) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] Re: GXP-2000 (steer clear)
On Wed, June 7, 2006 14:09, Louis-David Mitterrand said: On Tue, Jun 06, 2006 at 11:26:20PM -0400, Daniel Salama wrote: Well, these are encouraging words :) You're basically telling me that I should tell my client to buy other phones. I agree that you cannot compare these phones with Cisco or Polycom. After all, like you said, what do you expect for under $90. However, the fact is that my client just recently invested in these and it will be hard, if not impossible, for me to tell my client to swap them for Polycoms or something else at a much higher cost. I have heard complaints from my client about the speakerphone and they are now, I guess, getting used to picking up the handset :). I have heard any echo problems so far. What bothers me the most is that the phone stops working often (multiple times per day). By this I mean that my client won't be able to dial anything successfully. As soon as 3 or 4 digits are entered, they get a fast busy. To solve it, they need to reboot it. It sounds as if these phones were running Windows instead of Linux :) Anyway, what firmware did you use that solved so many of your problems? I've only had bad experiences with these phones and steer clear of them. In the same price range you can now get the Thomson ST-2030 or Polycom 430 for a much, much better user experience. Where do you purchase the Thomson or Polycoms for a comparable price as the GXP2000? I'd like to buy an ST2030 or 430 for under EUR 90 myself too! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] registration at Voipbuster times out
On Mon, May 29, 2006 16:20, Remko Muis said:
Hi Steve Attilla,
Thanks for the quick replies!!
Attilla: your suggestion sounds promising, since I know my system clock is
not too accurate. But that is the reason I use the network time protocol
daemon. Time and date settings are now correct.
Steve: your question about pinging the sip-proxy servers hits the nail on
its head: I can't, even though the names resolve to ip-addresses, and I
can
ping lots of other machines in the outside world. But why?
I tried your second suggestion, but to no avail. My dial statements were:
exten = _0[12345789],1,Dial,SIP/voipbuster-out/0031${EXTEN:1}
exten = _0[12345789],2,Congestion
exten = _XXX,1,Dial,SIP/voipbuster-out/0031[b]10[/b]${EXTEN}
exten = _XXX,2,Congestion
Replacing voipbuster-out with username:[EMAIL PROTECTED] does
not
help.
However, I did not really expect so, since the registration timeout errors
occur while Asterisk executes chan_sip.c. I would think that registration
fails independently of any wrong settings in extensions.conf.
Anyway, the s in the Contact-line does look suspect to me, since I have a
voip-in number for Voipbuster, and I read on the voip-info pages that the
s
extension is is used when there is no known called number in the context
used.
Being an Asterisk-newbie, I appreciate your replies, but further
suggestions
even more ...
Remko
Remko,
What IP's do you get returned for sip.voipbuster.com?
Do you use UU-net's DNS servers? If so, you might try using different
servers, as I have had some weird experiences with their DNS servers in
the past.
Have you tried trace-routing to the server to see where it breaks?
I am using voipbuster as well, and am usually able to connect just fine...
Good luck!
--
F Peeters
PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0
AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN
2 Sweex HFC-PCI cards
___
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Re: [Asterisk-Users] Call terminated after 60 seconds
On Fri, March 24, 2006 12:01, Asterisk said:
Hello,
I switched from my PSTN provider to a voip provider. (Voicedata in
the Netherlands)
From the moment i switched all inbound calls are terminated after
aproximatly 1 minute.
The provider tells me it's not their issue since I have no other
configuration than all their other users.
What can I do.
I removed all asterisk functionality by forwarding the inboud call
directly to a local phone
; Inbound voicedata context
;
[from-voicedata]
exten = ${VOICEDATACIDNUM},1,NoOp(From Voicedata)
exten = ${VOICEDATACIDNUM},n,Dial(SIP/2200,45,tr)
; end of context
Regards,
Andre Vink
Check whether your firewall has a fixed UDP timeout set at 60 seconds...
That solved my problem... ;-)
(Together with activating SIP/VoIP support)
--
F Peeters
PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0
AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN
2 Sweex HFC-PCI cards
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Re: [Asterisk-Users] Zap--IAX codec?
On Tue, March 21, 2006 16:51, Mimmus said: Hi, at my Asterisk box, I have a few of IAX2 phones (configured with alaw/ulaw/gsm codecs, in this order) and a PRI E1 line. In iax.conf I hav: disallow=all allow=alaw allow=ulaw allow=gsm During some incoming call, I read at console: -- Executing Dial(Zap/2-1, IAX2/215|20|TtwW) in new stack -- Called 215 -- Call accepted by 10.97.1.7 (format ulaw) -- Format for call is ulaw -- IAX2/215-33 is ringing -- IAX2/215-33 answered Zap/2-1 Why I have 'Format for call is ulaw'? I'd like to have alaw but keep ulaw to accomodate errors in various configurations (if any, not here!). EuroISDN uses uLaw, so Asterisk does as well, because it doesn't need to do transcoding then... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] Zap--IAX codec?
On Wed, March 22, 2006 0:06, Steve Kennedy said: On Tue, Mar 21, 2006 at 10:57:06PM +0100, Francesco Peeters (Asterisk) wrote: Why I have 'Format for call is ulaw'? I'd like to have alaw but keep ulaw to accomodate errors in various configurations (if any, not here!). EuroISDN uses uLaw, so Asterisk does as well, because it doesn't need to do transcoding then... Err,, uLaw is used by North America (as in U(s)Law ;) aLaw is used in Europe and other sensible areas. Steve Oops, you're right... my Bad! Sorry! (It's been a very long and tiresome day yesterday... I should have just kept my mouth shut!) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] IAX choppy sound
On Thu, March 16, 2006 12:08, Stojan Sljivic - GDS said: Hi, Does anyone know what would be acceptable RTT. Is 200ms OK? Regards, Stojan Sljivic When any of my VPN tunnels get over 100ms I start to get worried! Avg speeds on the tunnels are below 45 ms... I guess it depends on the level of quality you're used to tho! (As well a how far aprt the networks are... Mine are all in the same country...) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] G.729 codec licencing
On Thu, March 16, 2006 22:38, rnacharya said: Hi.., we have two asterisk server interconnected to each other through IAX2 trunk in two separate office. with this bellow configuration do we need to have Licensing for using G729 codec Office A T1 - Astrisk TE05PIAX2Astrisk Box -2 | | | | | | EPBX-1 EPBX-2 | | | | Telephone Telephone Thanks. Rudra. Your information is too summary to be able to tell... If EPBX-1 and -2 do G729, and the (*) servers only pass it, then you won't need additional licenses. Unless the (*) servers need to handle voicemail from either side. If the (*) servers have to transcode from any other codec *OR* from analog or ISDN (uLaw/aLaw) then you'll need licenses allright... Your best bet may be to contact Digium and give them all the details they need to determine the correct # of licenses... Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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: Over 40 destinations for FREE!]
Just in my Inbox: Original Message Subject: Over 40 destinations for FREE! From:[EMAIL PROTECTED] [EMAIL PROTECTED] Date:Thu, March 2, 2006 17:40 To: -- Dear Voip-Fan, From the makers of Voipbuster: http://www.internetcalls.com Over 40 FREE destinations, PLUS free VoipIn number AND Call Forwarding! For more rates, click here: http://www.internetcalls.com/en/rates.html Kindest regards, The VoipBuster Team If you want to be removed from our mailing list click here: http://www.voipbuster.com/en/feedback.html ___ --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: Over 40 destinations for FREE!]
On Thu, March 2, 2006 18:26, trixter aka Bret McDanel said: On Thu, 2006-03-02 at 17:51 +0100, Francesco Peeters (Asterisk) wrote: Just in my Inbox: From the makers of Voipbuster: http://www.internetcalls.com Over 40 FREE destinations, PLUS free VoipIn number AND Call Forwarding! Finerea has sipdiscount.com which also is offering the same deal. it appears they have peaked now and are mailing everyone off all their family of sites. I got one a while back for um something other than voipbuster I forget which of the 10 companies they operate (all basically the same deal). sipdiscount still makes you sign up with their stupid windows client but it freely gives you the sip settings so you dont have to guess if its sip.voipstunt.com or connserver.whatever or ... my guess is they are deprecating the other sites soon becuase they seem to really want to push internetcalls.com ... With all the sites integrated in to a single set of servers, and apparently the only difference between all service being the username, I'll stick with VoipBuster as long as I have credit... (My 120 days are passed, but my account and credit are still there... Maybe because I purchased before the expiration bit came in to play. (Might have to do with the many laws in Europe that do not allow for conditions to be changed *after* the purchase has been confirmed)) When the credit is almost gone, I'll check the situation again... ;-) The only thing I really miss is the free US calls... As long as most of Europe is free (esp The Netherlands and - in lesser extend - Belgium (Which was recently added back to VB!)) I am content... BTW: internetcalls.com has (currently) more free destinations than both VB and SD!... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] Random Disconnects - or ARE they?
On Wed, February 15, 2006 22:35, Brent Torrenga said: I have one use on our PBX who has been experiencing seemingly random disconnects. The user is on the same LAN as everyone else, using the same type of phone (79XX loaded with SIP firmware) as everyone else. He had some disconnects a few weeks ago, I suspected the phone, so I swapped his with mine. I have since not had issues with his old phone, however, he has had issues using mine. So, the problem seems to be not with the phone, but with his station. I started thinking maybe the cable is bad. I checked the network stats on his 79XX, and never see any receive errors - perfect network performance. Also, the CLI has no indication of an error whenever a disconnect occurs, it just looks like a normal hangup of the Zap channel (TDM400P). The ONLY difference between this user and everyone else is his extremely loud talking. When I run ztmonitor it is obvious that he simply pegs the meter. Either it reads peaked out or silence, whether he is speaking or being quiet. Is it entirely possible that he is driving the Zap channel so hard that it either hangs up or causes the telco CO to hang up the channel? Is there something else I should look at that might indicate what the problem is? I am kinda pulling my hair out on this one, any help or suggestions would be appreciated. LOL... You could try to explain that he doesn't need to shout to the person on the other side, that the telephone transmits the sound by wire, and not by air, so he doesn't need to shout to be heard on the other side! ;-) But seriously, I am really curious whether there is a connection between voice volume and disconnects... Please do keep us informed... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Problem with ZAPHFC: internal S0 hangs when hanging up
On Tue, February 7, 2006 9:53, Sven Fischer said: Am Dienstag, 7. Februar 2006 09:38 schrieb Sven Fischer: Hello all, if I try to call from one phone on the internal S0 to another on the same S0 using zaphfc, the bus is hung up. The called phone is ringing, but I can't talk from one phone to the other. The error I get is: -- Executing Dial(Zap/2-1, ZAP/1/55|15|tr) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called 1/55 -- Channel 0/1, span 1 got hangup, cause 42 -- Zap/1-1 is circuit-busy -- Hungup 'Zap/1-1' The called phone is still ringing, if I have hung up the calling phone. I have to restart asterisk to get things going again. Calling from SIP to the phones and calling from phones to external ISDN is working fine. Okay, further investigations show that if I connect just one phone to the NTBA, everything seems to work fine. If I plug in the second phone, the communication fails. Each phone works if plugged in on it's own into the NTBA. Termination in the NTBA should be activated, the switches are on. Where should I look for errors? Can it be a termination problem if every phone works on it's own? Sven Is the card set up for multipoint use? (BRI_NET_PTMP) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] 1 ISDN BRI to IAX2/SIP... (*) best tool or?...
On Tue, February 7, 2006 11:16, Peer Oliver Schmidt said: Francesco Peeters (Asterisk) schrieb: They have several ISDN BRI connections, most of which will be dropped. Only one will be retained, for 2 reasons: 1) It has the ADSL link 2) The number has been the main contact number for over 20 years. In germany you could move that number to a VoIP provider and use it from the main office direct. Then you won't need an asterisk in the remote location. Over here we can as well, but that requires cancelling the line it is on. That would mean we'd also lose the ADSL, and that would mean paying a penalty, paying connect fees all over again and then restart the entire provisioning circus all over again... My question is whether there are any tools better suited for this than an old banger (AMD 800 MHz) PC with a HFC-PCI card and (*) relaying (switch) the incoming calls to the central box. Should be plenty enough. I am running a PII-400 with a AVM C4 connected to two ISDN-ports and have another IAX connection to a customers site. Works fine. I have a PII-450 at home with 2 HFC-PCI cards (1 TE, 1 NT) with a few ISDN-DECT phones and a few IAX phones, which runs great. The only drawback is that starting AGI scripts takes a bit, so in and out bound calls take a bit longer to connect (10-20 seconds...) What I *also* would like to know is whether there's tools that people think would be better suited for this... IMHO a simple (*) box is the cheapest solution available, but I am always interested in novel ideas... ;-) -- F Peeters PII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] 1 ISDN BRI to IAX2/SIP... (*) best tool or?...
I have a question, I have to provide a solution for an office that will be almost abandoned, and there will be one or sometimes two persons 2 days a week. The main number however should be preserved. They have several ISDN BRI connections, most of which will be dropped. Only one will be retained, for 2 reasons: 1) It has the ADSL link 2) The number has been the main contact number for over 20 years. What we are looking for is to put a single SIP phone in the office, and have it connect back to an (*) server in the central office, where all other servers are located as well. In the remote office a single machine should be placed to terminate the BRI connection and relay it to the (*) server in the central office. That way the old number can be retained and an active phone can pick up the line as necessary. The preferred protocol to use would be IAX2, obviously. My question is whether there are any tools better suited for this than an old banger (AMD 800 MHz) PC with a HFC-PCI card and (*) relaying (switch) the incoming calls to the central box. (No intelligence there, no AGI scripts, just encode and transmit. Also no phones would need to be logged in to that machine, and outbound calling would only take place in very rare cases when the lines *and* VOIP connections at the central site are all congested...) TIA! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] RE: Euro-ISDN
On Wed, February 1, 2006 22:12, Armin Schindler said: On Wed, 1 Feb 2006, Aldo Bergamini wrote: [EMAIL PROTECTED] is believed to have said: chan_capi does not set the NT-mode. Your cards driver need to do that. E.g. for Eicon DIVA Server cards, you just set the '-x' option with divactrl or set NT-mode in the config wizard. chan_capi does not (need) to know anything about what protocol the card is doing. CAPI is independent here. Ok. Anyway, if the card is set to NT mode, you should specify ntmode=yes in the capi.conf to tell chan_capi to handle the progress better (get progress tones). Fine! One last related subpoint: while Eicon Diva cards have their own setup application, is there anything standard to control the basic setup of generic HFC-S cards? (something similar to the ztconfig tool for analog cards) Sorry, I cannot answer that one. I don't know enough about these cards and their drivers. With BRIstuff you get to use ztcfg, etc. Cannot say anything about mISDN, CAPI... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] meetme and dtmf
On Fri, February 3, 2006 0:44, Imran Ahmed said: Step 3 The Iax client heve to send some other DTMF to the IVR. How is the IVR still involved if the call has been transferred into a conference room? The IVR records the conversation between the other partecipant to the conference and wait '#' to stop recording and a '1' to save the file. may or may not work, try at your own risk: 1) Use a sip soft phone and set the dtmf mode = inband. 2) In asterisk set the dtmf mode for that soft phone to be rfc2833 or info. (this is done so that asterisk ignores the inband dtmf on the sip channel). 3) Design your dialplan such that asterisk should not depend on dtmf from the sip call. ex: exten xxx, 1, dial(zap/g/client_number) //on answer directed to conference room exten xxx, 2, dial(zap/g/ivr_number) //on answer directed to conference room. exten xxx, 3, meetme(conference room) once the sip call is in the conference then the ivr will detect dtmf from the audio data. Note that before the sip call is in a conference dtmf will not be detectable by the ivr or asterisk, and Ofcourse, this is not tested and only a test can confirm if it works. drawbacks: dtmf will not be available to ivr until your call is in conference. asterisk will never see any dtmf (which should be okay in this specific case). dtmf tones are not squelched so the other user in the conference will hear dtmf tones. Imran What I find strange is that the meetme IVR participant *does* hear DTMF from the ZAP channel, but not from the IAX2 channel... There shouldn't be a per channel difference in how dtmf is handled in meetme, right?... Do you know whether the IAX2 dtmf is intercepted by meetme and handled internally? If so you might be able to workaround by using SendDTMF() in your meetme dialplan... Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] meetme and dtmf
On Wed, February 1, 2006 12:07, Accursio Avona said: Imran Ahmed wrote: Here is my problem, at this point the IVR doesn't hear the dtmf sended by the iax client, even if it can hear the dtmf sended by the first zap channel. I donot know if IaxComm has inband dtmf mode available, if so enable it and see if it works. Someone can suggest me a Iax softphone with inband dtmf mode available ?? Thank's in advance AFAIK there's no DTMF option in IAX2... IAX always sends DTMF inline, eliminating the confusion often found with SIP. http://www.voip-info.org/wiki-IAX -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] meetme and dtmf
On Wed, February 1, 2006 15:04, Accursio Avona said: Francesco Peeters (Asterisk) wrote: SNIP AFAIK there's no DTMF option in IAX2... IAX always sends DTMF inline, eliminating the confusion often found with SIP. http://www.voip-info.org/wiki-IAX If so, wy the IVR does not hear the dtmf sended by the iax client and it hear that one sendee by the zap channel? Could it be a meetme problem? and if so what can i do? Thank yuo very much for any help. Accursio Avona Are you sure it *is* sending DTMF in the first place? (Just trying to find a logical place to start here...) I do not use meetme, but when I use idefisk, my (*) server recognizes the DTMF. Have you tried whether the IAXCOMM DTMF is recognized OUTSIDE meetme? -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Voipbuster incoming
On Tue, January 31, 2006 14:35, bails said: Hi all, Some friends of mine have an asterisk box which they use for outgoing IAX2 via voipbuster.com. They have been told that they now have an incoming number 0044117*** The thing is I cant seem to get any debug info on the incoming. I have tried both sip and IAX trunks but dont see any incoming info. Anyone have any idea what protocol voipbuster use for incoming calls?? Thanks in advance VB incoming ONLY works with SIP, not IAX2, which will be obsoleted shortly anyway. Incoming context will be the default SIP inbound context Incoming DID will be VB username My (working!) config: [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ilbc allow=gsm allow=g726 allow=speex allow=ulaw allow=alaw context = from-trunk ; Send unknown SIP callers to this context callerid = Unknown register=telno:passwd:[EMAIL PROTECTED] [username] allow=ilbcgsmspeexg726alaw ;currently only G728 and aLaw supported auth=md5 canreinvite=no context=from-pstn;seems to be ignored :-( disallow=all dtmfmode=auto fromdomain=sip1.voipbuster.com fromuser=username host=sip1.voipbuster.com nat=yes qualify=1000 realm=sip1.voipbuster.com secret=XXX type=friend username=username HTH! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Interface card for Euro-ISDN (BRI)
On Tue, January 31, 2006 10:43, Juergen K. Zick said: HI, all newer HFC-S cards will do. Depending on your application and system, you could easily ebaying an used Fritz!Card PCI or some active AVM B1 controller. Depending on the card you want to use you must se ZAPHFC or mIISDN/chan_isdn or chan_capi or mixtures with 2 different cards ... good luck, but there are enough HowTos available ... --Juergen For HFC-S cards you can also use vISDN!!! It supports TE and NT modes... It's still a bit immature (jitter and echo need work) but showing great promise! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] OT?: International number parsing
On Fri, January 27, 2006 23:47, Script Head said: What you're trying to accomplish can be easily done with an SQL query. You need to create a table of all the prefixes (international dial+country code+city/carrier) and join by that prefix. On 1/27/06, Damon Estep [EMAIL PROTECTED] wrote: Can anyone shed some light on rules that might make the task of parsing the country code and city codes from a dialed number in the CDRs? I know that there is almost never a case where a concatenated country and city code could overlap with another country code, but what about city codes and local numbers? Is it possible for a concatenated city code and local number to match another city code in the same country? I already have the table of country and city codes built. Are there holes in this theory; 1. Starting after the international dialing code, find the longest match for country code. 2. Starting after the country code from step 1, find the longest match for city code within that countries table of city codes. 3. The rest is the local number. Are there known exceptions? Am I reinventing the wheel rather than finding the right already existing resource? Obviously countrycodes are unique, and are created in a few 'classes' which also always provide unique numbers. Only one country has a single digit code: USA = 1 Most countries have a 2 digit code (31 = NL, 44 = UK, 49 = DE, etc.) There are *no* country codes with more than two digits that overlap the 2 digit codes. (So there's no 3 digit CC that starts with, for example, 31, 44, 49, etc.) So it is possible to 'categorize' them in to 1, 2, 3 digit CC's. Also the international dial codes have been chosen to not overlap anything else. So if you see (for instance) 011 you will always know it is an international call, and the next 1-3 digits will be a country code. -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] AAH out bound routing problem
On Fri, January 27, 2006 15:13, ram said: Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700 -- Got SIP response 488 Not acceptable here back from (PeerIP) -- SIP/easycall-838e is circuit-busy ram Most likely the telno provided (19197543700) is not compatible with what they expect... Maybe you need to att digits (Perhaps 0019197543700) or remove digits? Or maybe you're not authenticated ? We'll need more info to be able to assist any further... To begin with it would help to know what configuration they expect... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] External IAX2 phone defined as internal behaving as from PSTN
On Fri, January 27, 2006 16:09, Ian Cowley said: Have [EMAIL PROTECTED] 1.2.1 The server is on an internal network eg 10.10.10.10 It is NAT'd 1:1 via Checkpoint firewall to external public IP eg 50.50.50.50 The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on extension 1055. Outbound calls to 1055 work perfectly. Inbound calls from 1055 get picked up as if it were an external call (see below) and goes straight to the ring group macro. The same phone either on the same internal network to the asterisk or on a VPN to said network work fine. Obviously asterisk thinks this call is external. How do change this? SNIP The actual iax.conf part pertaining to this phone might be helpful here... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] External IAX2 phone defined as internal behaving as from PSTN
On Fri, January 27, 2006 17:23, Ian Cowley said: Iax.conf [general] ;bindport = 4569 ; Port to bind to (IAX is 4569) bindport = 5036 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=g729 ; 4 simultaneous allowed allow ilbc ; prefered for iax2 allow=gsm ; 13 Kbps (full rate), 20ms frame size allow=ulaw ;(g711)64 Kbps, sample-based allow=alaw ;(g711)64 Kbps, sample-based mailboxdetail=yes jitterbuffer=yes context=from-internal #include iax_additional.conf #include iax_custom.conf iax_additional.conf [1055] username=1055 type=friend secret=# record_out=Adhoc record_in=Adhoc qualify=yes port=4569 notransfer=yes [EMAIL PROTECTED] host=dynamic context=from-internal callerid=device 1055 Regards ianC Looks like you are using AMP / [EMAIL PROTECTED] As far as I can tell, this should work correctly... There might be something going on in the translation by the Checkpoint NAT control... Have you tried iax2 debug to see what it is receiving? the first few packets should give you sufficient information... Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Voipbuster problem
On Tue, January 24, 2006 12:09, RumaTech said: Hi, all I have a problem using voipbuster (and voipstunt) for that matter. On all calls, voice is disconnected after 30s. Asterisk still thinks that call is in progress and I do not get any tones, just silience. Remote party gets normal tones for disconnection. I have paid my 10e, so it is not that. Technical support bever came back to me. I have used them before on IAX, now I am running SIP. Same here: IAX2 worked fine, SIP now works sometimes, partially and unreliably! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Installing the none commercial intel g729codecs into [EMAIL PROTECTED] 2.2?
On Sun, January 22, 2006 13:02, Charles Wang said: I have the same problem too. I install the G.729 (IPP) to asterisk 1.0.x, and it works well. When I change asterisk from 1.0.x to 1.2.x, and G.729 seems work fine. I can use show translation and find it too. But when I make a call using G.729. The asterisk (1.2.1) crashed. If i mark the line allow=g729 from sip.conf. And asterisk works fine. Just tested with 1.2 trunk to another 1.2 machine with g729, and all worked fine! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Installing the none commercial intel g729codecs into [EMAIL PROTECTED] 2.2?
On Sun, January 22, 2006 19:40, Douglas Garstang said: Hang on there's a non commercial G729 codec that will work with Asterisk? Can someone point me to where I can find it? Thanks, Doug. Intel provides a sample for non-commercial/testing. http://www.voip-info.org/wiki-ITU+G.729 and http://www.voip-info.org/wiki/index.php?page=Asterisk+G.729+pass-thru The latter also has a link to the binaries... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Is sip1.voipbuster.com corking reliably for others on list?
On Sun, January 22, 2006 22:32, Ron Wellsted said: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Guillermo Salas M wrote: I've the same problem with sip1.sipdiscount.com. The calls are not connecting but are billed. SIPDiscount seem to have been having intermittent problems since Friday morning. It seems to be working now however. Will be testing again tomorrow! ;-/ -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Is sip1.voipbuster.com corking reliably for others on list?
I am trying to move from IAX2 to SIP for voipbuster, moving at the same time to sip1.voipbuster.com. When I try calling out, I see that there is SIP exchange, and in many cases also RTP data being exchanged. Hover in a very large number of attempts the connection is not established. Half of the time there is no RTP, the rest of the time there *is* RTP data flowing in two ways, but no ringtone is heard, and after a while the connection is terminated... Before I put in more time to investigate this, I should like to ask if people in general have any (good?) experience with VB's new SIP servers?... TIA BRgds -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2?
On Sat, January 21, 2006 22:10, MapsAir said: Has anyone successfully Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2? I tried to follow the instruction from http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ and http://aussievoip.com.au/tiki-index.php?page=G729-Install but I can't. I did it with [EMAIL PROTECTED] 1.5, but not 2.2 Working on it now... Will let you know how, if I succeed! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2?
On Sat, January 21, 2006 23:21, Franz Bräuer said: Hi, MapsAir wrote: Has anyone successfully Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2? Installed them today. Installing from source didn't work for me (Debian, Asterisk 1.2 from svn) but just adding the binaries (see the wiki on voip.org) did the job. Have you already tried the binaries? Kewl! Those work like a treat! As my testbox is a PII-750 running [EMAIL PROTECTED] 2.2 I did: cd /usr/lib/asterisk/modules/ wget http://kvin.lv/pub/Linux/Asterisk/codec_g723-gcc-pentium2.so wget http://kvin.lv/pub/Linux/Asterisk/codec_g729-gcc-pentium2.so After reloading, 'show translation' gives: Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 -22 8 817 8 724 115 19897 gsm 151 - 7 716 7 623 114 19796 ulaw 14616 - 111 2 118 109 19291 alaw 14616 1 -11 2 118 109 19291 g726 154241010 -10 926 117 20099 adpcm 14616 2 211 - 118 109 19291 slin 14515 1 110 1 -17 108 19190 lpc10 161311717261716 - 124 207 106 g729 16939252534252441 - 215 114 speex 16030161625161532 123 - 105 ilbc 17343292938292845 136 219 - Jolly good show, old chap! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] AIX calls with sipdiscount
On Fri, January 20, 2006 21:46, Roberto Pereyra said: Hi Someone have luck using Sipdiscount service with IAX ? I only can use sipdiscount IAX service using a free account (only 1 minute call) , I have a normal account and with it can login in the IAX server. I using sip1.sipdiscount.com like IAX server but can make free calls (less 1 minute). Thanks in advance. roberto Finarea s.a. are discontinuing IAX, soon! So it's not worth the effort to try to make it work! Only iax.* / sip.* (same host) does IAX2. sip1.* is apparently an outsourced server which only supports SIP. conectionserver1.* is the server to which their own client connects. Not sure what exact protocols are involved there! HTH! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Fritz card technology German *
On Thu, January 19, 2006 0:13, Hans Witvliet said: On Wed, 2006-01-18 at 11:45 +, John Daragon wrote: snip You can't use a Digium card because Digium doesn't make an ISDN2 card. snip If i see how many questions/complaints there are on the list about isdn/bri i would allmost wonder why digium does not make a single/quad active bri board Bri may not be popular as PRI in the usa, here in NL it's quite the opposite. PRI is way off limits for SOHO: it costs an arm and a leg initially and several toes a month ;-) I hear ya! We're using several BRI's rather than a PRI. We do not need the full complement of channels a PRI offers, but if prices were more reasonable we might have considered it anyway, simply because 1 PRI is much easier than several BRI's. Prices are so outrageous though that we settled for multiple BRI's and take the extra hassle for what it is... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Fritz card technology German *
On Tue, January 17, 2006 22:10, Camilo Gonzalez-Cortes said: The Fritz cards was not designed to run on asterisk whereas the following German ISDN cards (http://www.junghanns.net/en/quadBRI_produkt.html) was designed specially to run on this platform. The only problem with this vendor is the support...It is terrible. They never respond an e-mail Almost any card with the cologne HFC-S chip will work with their drivers + Florz patch, mISDN or vISDN. In my epxerience vISDN gives the best EURO-ISDN support, but it is a very young project, and still misses crucial stuff like echo cancelling... It is moving at a high pace though, so keep an eye on it... BriStuff is the most mature, but also still has bugs, and contrary to the vISDN developer, they hardly ever respond to emails... Whatever you choose, good luck! :-) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 - vISDN 2 Sweex HFC-PCI cards ___ --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] Re: automon - one touch record
On Fri, January 13, 2006 8:51, Tomislav Parcina said: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Also: What are the SIP CanReinvite settings for these phones? This shuldn't be important because he have w and W in his dial plan. * doesn't allow reinvite if you have t, T, w or W. It shouldn't make a difference, but should not and does not isn't always the same thing! I like to be thorough and systematic when problem solving... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Re: Re: automon - one touch record
On Fri, January 13, 2006 13:29, Tomislav Parcina said: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... It shouldn't make a difference, but should not and does not isn't always the same thing! We can't discus about this topic. It is simply meather of opinion. You think that is important and I don't. I like to be thorough and systematic when problem solving... Me to, that why I dont bother with erelevant things and care only about things that are relevant. Like I said before, it is mine and your opinion. It has no point discusing about it. In other words: Let's agree to disagree! ;-) That is fine with me... Have a nice weekend! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] AMP and additional conf files
On Thu, January 12, 2006 19:18, Ben Ferguson said: Hello all. I've been searching and can't quite find what I'm looking for... I've gotten AMP installed and up and running quite decently on an Asterisk box and am now in the process of tweaking it to my needs. My company currently has around 70 employees and we are running on a complete Avaya system, but this system is no longer going to work for us (too much money for not enough stuff). So I have been put in charge of setting up an Asterisk PBX and get an entire test system going on it to see if Asterisk will meet our telephone needs. Extensions, queues, voicemail, stats, etc etc. Here's the problem: this Asterisk server is actually currently running live, serving information to people calling in to it. I need my test office setup, with AMP and this other system to work simultaneously, but yet totally separate. As my stuff is for a test, I would like to set it up so that when I dial in TO my Asterisk PBX FROM a specific telephone number, it takes me to my office test section in asterisk, otherwise, from ANY other number, it dials the info serving section. This would allow me to call from a certain telephone number and be able to get to my test office setup, but if anybody else calls from any other number, they get the other stuff. Doesn't sound too bad right? So how would one do this using AMP if AMP is more of the secondary system? If I understand correctly, to add additional, custom contexts to extensions.conf, it should be entered into extensions_additional.conf and the contexts should contain the word custom in them. So, first question, what if I want that custom context to be the first context (as in possibly the default context), but only if it's from a certain telephone number...? I assume you would enter that custom context as the context in zapata.conf, but how would you tell it to go back to the AMP stuff if the FROM telephone number is my speicifc telephone number? What context would I send it to so that it will do the regular AMP stuff? (Incidentally, I have a local telephone number and an 888 telephone number coming into my PRI, but when called, my Asterisk PBX views/receives them both as the local telephone number.) SNIP Normally in AMP (depending on version) you'd make either an inbound route like this : 4081234567|4081234599 (where the 4567 is the DID and 4599 the callerID) or an inbound route with DID=4081234567 and CID=4081234599 and then send it to a specific extension or custom context... HTH -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] automon - one touch record
On Fri, January 13, 2006 5:15, Jennifer Hales said: Hello all, I am unable to get automon recording to work; can someone advise me what I am doing wrong? When I do *1 all I see in the CLI screen is attempting native bridge of SIP/3006-291b and SIP/3153-6fdd, and there is no call record generated in /var/spool/asterisk/monitor/. Here are my settings: SNIP Does transferring with # or *2 work? (Or whatever sequences you assigned to those functions in feastures.conf...) That way you can get an idea whether it is just automon, or whether there's a more generic issue... Also: What are the SIP CanReinvite settings for these phones? Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Transfer sounds - notifications
On Wed, January 11, 2006 12:46, Tomislav Parcina said: When I try to make attendend transfer (*2) this what hapends. I press *2 other person goes on hold and I hear transfer. I press extension number and that extension starts to ring but I don't hear anything. If nobody picks up that phone call in few seconds I get back to the person I was talking to (the person I triesd to transfer). The problem is that again, I don't hear anything (that person waits for me to say something) and I don't know that I'm back to transfered person. I hope that I have make it clear enough. Anyway, how can I solve this one? I would like to hear that the phone of extension is ringing, and I would like to konw when I'm speaking again with my caller. On http://www.voip-info.org/wiki-Asterisk+config+features.conf: ;courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call ;xfersound = beep ; to indicate an attended transfer is complete ;xferfailsound = beeperr; to indicate a failed transfer You could try these to see if that makes a difference?... Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] IAX CallerID
On Wed, January 11, 2006 7:52, scott said: Hi All Apologises if this has been disussed and I missed it. My SetUp I have a sip phone registered to an asterisk box (a1) in one location 1. This phone dials an extension which is in another location, so a1 passes the call via IAX to the other asterisk (a2) in location 2 which then dials the local phone. My Problem The caller ID setup in the sip.conf for the phone registered to a1 is not passed via the IAX to a2 and is therefor not being displayed on the phone in location2. The only way I can get the phone in location2 to display the caller ID is to set the callerid in the user part in the iax.conf on a2. Hope this makes sense Many thanks It sure does, as I am examining exactly the same issue for my set up... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] IAX CallerID
On Wed, January 11, 2006 16:00, Colin Anderson said: As a rule of thumb, I always explicitly set CallerID in my dialplan before making a call through IAX, SIP or PSTN. If you make it part of a generic dialout routine then it isn't a hassle. It always works. It sometimes doesn't for my installation, but I'll check it later, it is not a big issue right now... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Zaptel modules load, but Asterisk fails at startup
On Wed, January 11, 2006 19:35, Stephen Bosch said: I'm running Asterisk on a Gentoo box with the Zaptel 1.2.1 drivers. If I boot the machine without having the wcfxs module autoload, then install the module with modprobe, asterisk works just fine. If I boot the machine and autoload the wcfxs module, the module loads fine: Jan 11 11:06:55 asterisk Zapata Telephony Interface Registered on major 196 Jan 11 11:06:55 asterisk ACPI: PCI Interrupt Link [LNKC] enabled at IRQ 10 Jan 11 11:06:55 asterisk PCI: setting IRQ 10 as level-triggered Jan 11 11:06:55 asterisk ACPI: PCI Interrupt :00:0a.0[A] - Link [LNKC] - GSI 10 (level, low) - IRQ 10 Jan 11 11:06:55 asterisk Freshmaker version: 73 Jan 11 11:06:55 asterisk Freshmaker passed register test Jan 11 11:06:55 asterisk Module 0: Installed -- AUTO FXS/DPO Jan 11 11:06:55 asterisk Module 1: Not installed Jan 11 11:06:55 asterisk Module 2: Not installed Jan 11 11:06:55 asterisk Module 3: Installed -- AUTO FXO (FCC mode) Jan 11 11:06:55 asterisk Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules) The module is running: asterisk sfbosch # lsmod Module Size Used by wctdm 39936 - zaptel226756 - asterisk sfbosch # But Asterisk behaves as though it were not: [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Jan 11 11:32:53 WARNING[5778]: chan_zap.c:920 zt_open: Unable to specify channel 1: No such device or address Jan 11 11:32:53 ERROR[5778]: chan_zap.c:6847 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Jan 11 11:32:53 ERROR[5778]: chan_zap.c:10251 setup_zap: Unable to register channel '1' Jan 11 11:32:53 WARNING[5778]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 Jan 11 11:32:53 WARNING[5778]: loader.c:554 load_modules: Loading module chan_zap.so failed! Warning, flexible rate not heavily tested! asterisk sfbosch # Ouch ... error while writing audio data: : Broken pipe Looking at this now as I write this, it seems that some module dependencies aren't loading, but I can't be sure. Does anybody have an idea what's going on here? -Stephen- Try running ztcfg -vvv -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Zaptel modules load, but Asterisk fails at startup
On Wed, January 11, 2006 21:36, Stephen Bosch said: Francesco Peeters (Asterisk) wrote: On Wed, January 11, 2006 19:35, Stephen Bosch said: Try running ztcfg -vvv Yes, that fixes it -- my question, I guess, is how to get that to run automatically at boot time... -s Either put it in rc.local or in /etc/modules or /etc/modprobe.conf or whatever the equivalent is on gentoo For example in my /etc/modprobe.conf: install wctdm /sbin/modprobe --ignore-install wctdm /sbin/ztcfg alias wcfxs wctdm HTH -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Zaptel modules load, but Asterisk fails at startup
On Wed, January 11, 2006 23:37, Tzafrir Cohen said: On Wed, Jan 11, 2006 at 01:36:24PM -0700, Stephen Bosch wrote: Francesco Peeters (Asterisk) wrote: Try running ztcfg -vvv Yes, that fixes it -- my question, I guess, is how to get that to run automatically at boot time... I run ztcfg in a spcial init.d script for zaptel (which also does other clean-ups). Nothing stops you from running ztcfg in the asterisk init.d script. BTW: there is no point in the -vvv: ztcfg will be nice and verbose in reporting errors when they happen. No need for the extra noise (and wasted time) at boot. I agree about the -vvv being superfluous. I only added it to get confirmation that it actually had seen the card and it's ports in case it didn't work as expected... ;-) You may notice that there's no -vvv in the modprobe.conf sample lines either... Cheers! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Same Zap channel in multiple groups
On Mon, January 9, 2006 16:44, Patrick Conroy said: Does anyone know if it would cause problems to have the same Zap channel in multiple goups? So, for example, if I have two PRIs would the following work or would it cause problems: channel = 1-23 group = 1 channel = 25-47 group = 2 channel = 1-23,25-47 group = 3 I am just curious if anyone has set some thing like this up and how it worked out. Thanks, Patrick AFAIK group = 1,3 channel = 1-23 group = 2,3 channel = 25-47 should work... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Decent sub-$100 SIP phone.
On Tue, January 10, 2006 6:03, Dovid B. Asterisk Users said: Ken, I would tell the client that you offerd phones for under $100.00 and he didnt like them so now for a diffrent phone he will have to pay more. Also I have an 841 and for it works great. I also installed one for a customer in a mechanic shop and no complaints. Regards, Dovid I agree! They're the ones that don't want the 841. Also functionality is IMHO more important than looks, especially in an office/work environment. It'd be like getting a quote for a Suburban, then saying you don't like it and expecting an H2 for the same price instead... I would tell them that you'll need to requote for the phones... Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Recommendations on a WiFi phone for *?
On Tue, January 10, 2006 5:50, Ira said: At 05:44 PM 01/09/2006, you wrote: We're getting our feet more and more wet with VOIP at work. We want to experiment with a good wireless (as in WiFi) phone. What would be a good phone to impress my boss with? I have the Zyxel P2000W V2 and while it has it's user interface annoyances, it's a great little phone and only $150 if you look hard enough. The most annoying one is sleeping, I guess to save battery life but if you forget to wake it up it looses the first 3 or 4 numbers you punch in. But it worked perfect, the first IP phone I've ever had and once I figured out I had to put the WEP code in hex it registered and work perfectly, even had people tell me how good I sound. Zyxel to an * box out a TDM400 to a Linksys VOIP router to ATT Callvantage. Ira Another, much cheaper option is to get DECT phones and connect them to IAXy's: DECT-PHONE ((( * ))) DECT-BASEIAXy[=IAX2=]Asterisk- TheWorld(tm) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Recording Calls at the phone
On Fri, January 6, 2006 15:37, Michael Sampson said: I work for a call center and we are looking at using asterisk to have our operators take calls. Our message taking software records all the calls on the operators computers. Right now we use these recording controls from radio shack that plug in between the wall jack and the phone and plug in via a 1/8 inch stereo connector to the mic input on the computer. If I buy an IP phone I can't do that. I could get an FXO adapter and regular phones, but I'm looking to get as little equipment as possible. Radio shack makes a recording control that plugs in to a 2.5 mm headset jack, but it takes batteries so thats not going to work Does anyone else do something similar? Does anyone have any ideas about what producs/setup would work for this. Asterisk has a built in monitoring system. You can chose to do Always, Never or On Demand monitoring, depending on your setup and dialplan http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Monitor Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Announcing a call transfer
On Fri, January 6, 2006 15:46, Michael Sampson said: With our current pbx system, a call comes in from the PSTN to the receptionist. She then hits flash, which puts the caller on hold, calls my extension, says so and so is on the phone for you, I say ok put him through, she hangs up and I am connected to the caller. With [EMAIL PROTECTED] I can it # then the extension to transfer to and it will ring there. But is there a simple way to announce the call before you transfer it. If not, does anyone have any good work arounds for this. -- It is called attended transfer. See http://www.voip-info.org/wiki/view/Asterisk+PBX+functions And http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf HTH! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Not Able to Connect Two Asterisk Servers Using IAX2
On Fri, January 6, 2006 20:20, Chandan Mishra said: Hi I have two asterisk servers. I just want to connect two asterisk server using IAX2. But the Asterisk Servers are not able to register each other. If some body have done this then Please send me the configuration they have done in iax.conf and extensions.conf. I simply want to connect and call from one sever to another. Thanks Chandan Kumar Mishra Software Engg. As always, the Wiki is your friend... http://www.voip-info.org/wiki-Asterisk+-+dual+servers I am using a modified version of method 3... You have to make sure that you have a user entry in IAX.conf for the other server as mentioned above... So if your serverA logs in using passwd SECRET, make sure that you have an entry [serverA] secret=SECRET type=user context=IncomingContext auth=md5(this one is optional of course...) Good luck! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Anybody successfully using vISDN on [EMAIL PROTECTED]
Is there anybody in this group that is using vISDN on an [EMAIL PROTECTED] server? I have a couple of questions, which are quite lengthy, and I do not want to pollute this list of there's no use in asking to begin with! TIA BRgds -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ADM Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI ___ --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] Raw Hangup messages with IAX2?
On Wed, January 4, 2006 10:58, Matt Riddell said: Mike McMullen wrote: I found the problem. There was a misconfiguration in the person's firewall that once fixed cleaned everything up. Sorry for the wasted bandwidth. Just for curiosity's sake, what was the misconfiguration? I'd love to know too, as I too see these messages and would like to know how to prevent those... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Raw Hangup messages with IAX2?
On Wed, January 4, 2006 14:53, Mike McMullen said: Subject: Re: [Asterisk-Users] Raw Hangup messages with IAX2? Mike McMullen wrote: I found the problem. There was a misconfiguration in the person's firewall that once fixed cleaned everything up. Sorry for the wasted bandwidth. Just for curiosity's sake, what was the misconfiguration? -- Cheers, Matt Riddell Hi Matt, The person at home had their IAX2 ports forwarded to the wrong IP address. (Though they swore they didn't!) ;-) Mike Hmzzz... That's not my problem though, so I quess I'll need to investigate further! :-( Thanks for the info tho! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Re: Start recording after call started
On Wed, January 4, 2006 15:45, Tomislav Parcina said: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... In Asterisk v1.2.1 check the featuremap section of the features.conf file. You also need to add the w or W option to your Dial cmd where appropriate. So with the feature mapping below pressing *1 would start recording. [featuremap] blindxfer = #1; Blind transfer, default is # disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *2 ; Attended transfer I need to dail *1 to quickly. Can that be changed? Try experimenting with this: [general] featuredigittimeout = 1000 ; Max time (ms) between digits for ; feature activation. Default is 500 HTH! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. AMD Duron 1GHz - 1GB - * 1.2.1 2 Sweex HFC-PCI cards ___ --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] Asterisk 1.2.1 segmentation faulting!...
I am having issues with 1.2.1/BriStuff 0.3.Pre 1d/Florz patch
On a *very* regular basis I get:
Disconnected from Asterisk server
/usr/sbin/safe_asterisk: line 42: 1359 Segmentation fault
${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
Anyone seen this? Any ideas?
TIA BRgds
--
F Peeters
PIII 450 - 1 GB - * 1.2.1 - BRIstuff 0.3.0 Pre 1d - Florz patch
2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0
Cologne HFC-S pins #52, #54, #55 connected in parallel for synching.
___
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[Asterisk-Users] Outbound call using ISDN extension disconnected after *exactly* 30 seconds
Hello all, I have a curious issue, and I was hoping maybe somebody has an idea... I have a Siemens DECT ISDN base connected to a HFC-PCI card in NT mode. When I use it (or one of the connected DECT phones) pending outbound calls are disconnected after *exactly* 30 seconds (if the call is answered before that all works fine! It is only when the phone is still ringing that this fails!) When I use the base it reports 'Ongeldig' (Invalid) on the screen after disconnect. I have included the BRI INTENSE DEBUG output below, maybe someone has an idea what to look for. Also included is the config of the ISDN extensions and zapata.conf. I may be missing something totally obvious, but I am baffled, and this way it is unusable... Any thoughts will be appreciated! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. --- CLI output Supervisory frame: 2 SAPI: 00 C/R: 0 EA: 0 TEI: 064EA: 1 2 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 002 P/F: 1 0 bytes of data 2 -- Restarting T203 counter 2 -- Restarting T203 counter 2 terisk1*CLI [ 00 81 04 04 08 01 01 45 08 02 80 e6 ] 2 terisk1*CLI Informational frame: 2 SAPI: 00 C/R: 0 EA: 0 TEI: 064EA: 1 2 N(S): 002 0: 0 N(R): 002 P: 0 8 bytes of data 2 -- ACKing all packets from 1 to (but not including) 2 2 -- Since there was nothing left, stopping T200 counter 2 -- Stopping T203 counter since we got an ACK 2 -- Nothing left, starting T203 counter 2 Protocol Discriminator: Q.931 (8) len=8 2 Call Ref: len= 1 (reference 1/0x1) (Originator) 2 Message type: DISCONNECT (69) 2 [2 082 022 802 e62 ] 2 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) 2 Ext: 1 Cause: Unknown (102), class = Protocol Error (6) ] 2 Sending Receiver Ready (3) 2 terisk1*CLI [ 00 81 01 06 ] 2 terisk1*CLI Supervisory frame: 2 SAPI: 00 C/R: 0 EA: 0 TEI: 064EA: 1 2 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 003 P/F: 0 0 bytes of data 2 -- Restarting T203 counter 2 -- Restarting T203 counter -- Channel 0/2, span 2 got hangup request -- Hungup 'IAX2/voipbuster-4' == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'Zap/5-1' in macro 'dialout-trunk' == Spawn extension (from-internal, 0174287004, 1) exited non-zero on 'Zap/5-1' -- Executing Macro(Zap/5-1, hangupcall) in new stack -- Executing ResetCDR(Zap/5-1, w) in new stack Tx-Frame Retry[000] -- OSeqno: 009 ISeqno: 010 Type: IAX Subclass: HANGUP Timestamp: 22039ms SCall: 4 DCall: 00150 [213.61.187.146:4569] CAUSE CODE : 0 -- Executing NoCDR(Zap/5-1, ) in new stack -- Executing Wait(Zap/5-1, 5) in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'Zap/5-1' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'Zap/5-1' --- ZAPATA.CONF -- ; ; Zapata telephony interface ; ; Configuration file [channels] ; ; Default language ; language=nl ; ; Default context ; ; switchtype = euroisdn rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=10.0 txgain=0.0 nationalprefix = 0 internationalprefix = 00 faxdetect=incoming callgroup=1 pickupgroup=1 context=from-pstn ; PRI Out of band indications. ; Enable this to report Busy and Congestion on a PRI using out-of-band ; notification. Inband indication, as used by Asterisk doesn't seem to work ; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT ; inband: Signal Busy/Congestion using in-band tones priindication = inband ; p2mp TE mode ;signalling = bri_cpe_ptmp ; p2p TE mode ;signalling = bri_cpe ; p2mp NT mode ;signalling = bri_net_ptmp ; p2p NT mode ;signalling = bri_net pridialplan = dynamic prilocaldialplan = unknown nationalprefix = 0 internationalprefix = 00 echocancel=yes echotraining = 100 echocancelwhenbridged=yes signalling = bri_cpe_ptmp immediate=no relaxdtmf=yes overlapdial=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes group = 1,2,3,4 channel = 1-2 signalling = bri_net_ptmp priindication = outofband context=from-internal ;context=ext-local relaxdtmf=yes immediate=no overlapdial=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes group = 11,12,13,14 channel = 4-5 ;Include genzaptelconf configs #include zapata-auto.conf ;Include AMP configs #include zapata_additional.conf -- ZAP -- ;;[2010] signalling=bri_cpe_ptmp record_out=Adhoc record_in=Adhoc [EMAIL PROTECTED] echotraining=100 echocancelwhenbridged=yes echocancel=yes
Re: [Asterisk-Users] select codec based on extension
On Thu, December 29, 2005 9:52, Simone Cittadini said: Leandro Rzezak ha scritto: I'm having same problem. Were you able to solve it? No, codecs became a secondary problem later in our project so we ended up with 711 on all servers and more bandwidth, anyway the post refers to asterisk 1.0.something and I never investigated the problem in more detail... I think it's possible, usually when you receive no answers (as the case of that post) you have made a really silly question :) Either that or noone really knows the answer... ;-) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] Re: who is online
On Wed, December 28, 2005 16:38, bails said: qualify=yes in both sip.conf and iax.conf, seems to highlight both the users and trunks who are currently available in FOP Bails Note that some IAX clients do not seem to like qualify=yes. I use DIAX, and when I use Qualify=yes, it becomes unavailable after a while... Also see http://www.voip-info.org/wiki-Asterisk+config+iax.conf and scroll halfway down to the qualify header -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] Pls. explain what happens...
On Tue, December 27, 2005 9:26, Mauro Zanin said: Hi everybody, can anybody explain one thing: say we have 2 SIP phones(or H323) and one Asterisk Box on one local network. The phone1 calls phone 2 via Asterisk and phon3 answers: is the real conversation streaming thru the * box, or it's going straigth from one phone to the other? Regards and Happy New Year. Mauro That depends on several factors, but basically: CanReinvite = no = (*) always inbetween CanReinvite = yes = If no NAT or other limiting factors (firewalls, etc.) in place, phones will direct connect, otherwise (*) will handle the flow. HTH -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] Changing Automon filenames?
Hello all, Is it possible to change what filename automon (*1) files get, and if so, how? I checked the wiki, but only found info about filenames for normal monitoring. Does the same work for automon? TIA! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] Changing Automon filenames?
On Tue, December 27, 2005 13:49, BJ Weschke said:
On 12/27/05, Francesco Peeters (Asterisk) [EMAIL PROTECTED]
wrote:
Hello all,
Is it possible to change what filename automon (*1) files get, and if
so,
how?
I checked the wiki, but only found info about filenames for normal
monitoring. Does the same work for automon?
Not really. The only influence you get on the filename used here is
with the TOUCH_MONITOR channel variable. If that is set, the filename
format will then be auto-epoch time-${TOUCH_MONITOR}.formatext. If
you don't set it, the filename will then be, auto-epoch time-caller
chanid-callee chanid.formatext In any case, the channel variable
TOUCH_MONITOR_OUTPUT will contain the name of the file that was chosen
for the one touch recording.
--
Well, at least I *can* put information in there I want to have... It's a
kludge, but one that'll just allow me to do what I need :-)
Thanks!
--
F Peeters
PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0
Cologne HFC-S pins #52, #54, #55 connected in parallel for synching.
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Re: [Asterisk-Users] PRI: This number has been disconnected
On Tue, December 27, 2005 23:37, Javier Ergas said: Hi, I'm running [EMAIL PROTECTED] 1.5 with TE110P E1 PRI in Chile. When calling an invalid number using, I expect to hear: We're sorry you have reached a number which has been disconnected ... And that is indeed what I hear when I dial out from [*] using analog FXO, or VoicePulse or NuPhone. When I dial that same number trough the T1 / PRI interface however, I only hear the allison7/all-circuits-busy-now message. There was another issue like this in an old post (http://lists.digium.com/pipermail/asterisk-users/2004-April/043597.html) but I think it isn't the same. SNIP I believe this has to do with the AMP macro's being used in [EMAIL PROTECTED] I am seeing similar things. For instance: One issue I have is that when a route has multiple trunks, and the first trunk after a while returns with 'NOANSWER', it merrily continues to the next trunk, which is not quite the behavior I'd expect. Especially as the primary trunk (IAX/VoipBuster) is *much* cheaper (ie free) as compared to the second trunk (Zap/g1), but the switch is made without any message. This could mean that you might be talking to someone on a different trunk, and instead of a free call, be paying normal fees. This could become expensive if you're calling the USA from Europe!... I am currently looking in to ways to enhance those macro's to respond more reliably, as well as return more useful information (busy tone on busy and no-answer, number disconnected info, etc.) when needed. If I do get to a satifactory set of macro's, I will put them up on the Wiki and let the list know... (I'm just starting on doing manual configuring, so it will be a tough job to crack, but also a learning experience...) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] Merry Xmas to everybody!
On Fri, December 23, 2005 9:22, Mauro Zanin said: Hi everybody, no issues this time. Only stopped to say: Merry Christmas and Happy New Year. Ciao Mauro Same to you, and the rest of the list too! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] Re: How to record a call
On Thu, December 22, 2005 12:54, Tomislav Parcina said: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... For Asterisk 1.2: http://www.voip-info.org/wiki/view/MixMonitor Can this one be done on demand? Like, I dial *1 and it starts recording. http://www.voip-info.org/wiki-Asterisk+config+features.conf BTW: Please let me know when you've got this working 100%... I keep having issues with it! Most notably when dialling OUTBOUND with IAX softphone (tried borg DIAX and IDEFISK) Last time I checked, it worked for some of my DECT ISDN phones on ZAP (only the ones supporting 'dialpad mode') Looks to me like (*) has some issues with inband DTMF on outbound calls, but I need to test more before I can put together an exact description of the problem... (Next step is to test SIP phones with both RFC and inband DTMF) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] Re: Re: How to record a call
On Thu, December 22, 2005 15:27, Blake Krone said: I'm running AAH 2.2 and *1 works from my eyebeam sip phones to do on demand recording. Like I said SIP phones are next on the list to try! ;-) You need to set the DIAL_OPTIONS of wW in order to utilize this feature. lower case w means called person can initiate, upper case means callee can initiate, I think that is the order. Changed DIAL_OPTIONS in the database to read 'tTrwW' They show up as auto-timestamp-src-dst.wav in /var/spool/asterisk/monitor However, they will NOT show up in ARI, I modified the code to show them and sent the modification to Dan to implement if he chooses. -Blake Could you send me (off-list) the diff to look at? I am using AAH2.2 as well ;-) On 12/22/05, Tomislav Parcina [EMAIL PROTECTED] wrote: TIA! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] Re: Is it me, or is 1.2.1 slower than 1.0.9?
On Tue, December 20, 2005 9:13, Tomislav Parcina said: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Also, despite setting DYNAMIC_FEATURES=automon in the extensions.conf globals section and uncommenting automon=*1 in features.conf, nothing happens when pressing *1 Solved that... When I change blinsxfer in features.conf to anything different than #, it no longer works. That too... You can say what was the problem. I did: The Siemens ISDN Base was... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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 - Skype anywhere/anyhow?
On Mon, December 19, 2005 11:33, Evert Meulie said: Hi all! I am aware of products like http://www.rsdevs.com/psgw_sip.shtml which act as a gateway, but what I'd really like is a for example an Asterisk module that can route calls to Skype, perhaps the same principle as IAX2? I'm assuming more people are interested in this, but... does it exist already? There is no such thing yes, and as Skype is closed source, it'll have to wait until someone reverse-engineers it... (Sniffing the protocol will be hard, as it is - supposedly - encrypted) I'd love to connect my (*) to Skype as well, but I do not see it happening soon! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] Is it me, or is 1.2.1 slower than 1.0.9?
Hi all, I just wiped my system and did a clean Asterisk 1.2.0 install with Bristuff 0.3 Pre 1c. (It doesn't work with 1.2.1 yet!) :-( Is it my server or is 1.2.0 considerably slower than 1.0.9 was? It seems to me that all actions take noticably longer than before! Also, despite setting DYNAMIC_FEATURES=automon in the extensions.conf globals section and uncommenting automon=*1 in features.conf, nothing happens when pressing *1 When I change blinsxfer in features.conf to anything different than #, it no longer works. It only works with my softphones anyway, as my ZAP connected ISDN phone never transfers to begin with! I'm getting depressed, because I know all these nice features are there, and I cannot get any of them working! (Once it works, I can deploy it at 2 other locations and really start saving money...) Any suggestions? TIA! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1c - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] Is it me, or is 1.2.1 slower than 1.0.9?
On Sun, December 18, 2005 20:05, Francesco Peeters (Asterisk) said: Hi all, I just wiped my system and did a clean Asterisk 1.2.0 install with Bristuff 0.3 Pre 1c. (It doesn't work with 1.2.1 yet!) :-( Is it my server or is 1.2.0 considerably slower than 1.0.9 was? It seems to me that all actions take noticably longer than before! Also, despite setting DYNAMIC_FEATURES=automon in the extensions.conf globals section and uncommenting automon=*1 in features.conf, nothing happens when pressing *1 Solved that... When I change blinsxfer in features.conf to anything different than #, it no longer works. That too... It only works with my softphones anyway, as my ZAP connected ISDN phone never transfers to begin with! And here we come to the root cause: The Siemens ISDN DECT station stubbornly refuses to do DTMF unless I manually go in to a menu 2 levels deep to temporarily turn it on... No preference setting, etc. It even gets worse with non Siemens DECT handsets (using the GAP protocol), as these do not even support the keypad switching, which means I first have to do a DECT transfer to a Siemens handset or the base, before I can xfer to a non-DECT extension or external peer... I'm getting depressed, because I know all these nice features are there, and I cannot get any of them working! (Once it works, I can deploy it at 2 other locations and really start saving money...) So my depression is somewhat lifted, as I will be proceeding with the other 2 locations, as I now have all features working, however I need to figure out how to properly work around the Siemens issue, as the other sites too use Siemens ISDN hardware! :rolls eyes: Time to bring out the AMD 1000 box and start prepping that one! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] hint on Zap channels
On Fri, December 16, 2005 16:39, Kevin P. Fleming said: C F wrote: Kevin, I'm not sure this would work here, but maybe it would. There was a bug posted about being able to use hint against local channels, would that not help him? http://bugs.digium.com/view.php?id=5779nbn=4 No, the issue is that multiple ISDN devices are not distinct channels as far as Asterisk is concerned; they are all 'Zap/1' with different extensions behind that channel. This is the same question as asking 'if I have a PRI connected to my Panasonic PBX, can I use hints for all the extensions on that PBX'. It won't work in Asterisk, because it's not aware of the actual endpoints, only the channel that connects to them. I personally think this is a fault in (*). (Or rather Zaptel) Because there is such a thing as ISDN, I think it should be able to recognize separate channels for DIDs... Both internal and external ZAP channels should be able to recognise the different DID/CID/CLID as separate identifiable endpoints. That way you can chose a 'channel' and have (*) use the correct CID/CLID. When doing extensions it should dial it, when doing outbound the chosen channel could define which MSN/CLID to use, inbound the DID would define the channel. (Just like the way it does now for the channels/extensions, but for ISDN just dialing Zap/1 won't do the trick... You'll need to dial Zap/1/2020 to get the ISDN phone with MSN 2020) Just my EUR 0,02 -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] hint on Zap channels
On Fri, December 16, 2005 20:44, Kevin P. Fleming said: Francesco Peeters (Asterisk) wrote: I personally think this is a fault in (*). (Or rather Zaptel) You are certainly welcome to your opinion, but thinking that Asterisk should understand the concept of 'remote endpoints' as native devices is by no means a 'fault'. If nobody has wanted this enough before to be able to code it up and submit it, then it's just a lack of functionality. OK, Maybe fault wasn't the right word here... Lacking is probably better... I'd love to look in to it and code it, but I simply haven't got the time to investigate and code it... Because there is such a thing as ISDN, I think it should be able to recognize separate channels for DIDs... Both internal and external ZAP channels should be able to recognise the different DID/CID/CLID as separate identifiable endpoints. That way you can chose a 'channel' and have (*) use the correct CID/CLID. And how would Asterisk know when these endpoints communicate directly with each other to keep trace of device state? Because it would either be the device in NT mode, and therefore initiate the connection, and be able to see the data flows. Or it would be TE mode, but still on the same bus (which means it'll still see the data) It would certainly be possible to do what you want, but it would need to be implemented by the Zaptel driver that is communicating with that ISDN interface, so it can present distinct 'channels' to chan_zap for each device on the ISDN bus. That's why I said 'Or rather Zaptel' in my original comment... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] IAX2 show channels show Channel (NONE)
On Tue, December 13, 2005 13:47, Dmitry Zhukovski said: Hi all! I have got a bit strange output from iax2 show channels: Med venlig hilsen ComX Networks A/S Dmitry Zhukovski System developer Adding some info might be helpful? -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] Dialing analog extensions from SIP?
On Wed, December 14, 2005 3:33, Robert La Ferla said: Doug Lytle wrote: I agree with Eric on this one. On my Polycom IP501s, I had to change the digit map to allow for # and * matching. For testing, remove the # and try again. Remove it from the phone's dial plan or all together? Also, my phone has a local dial plan that is set to this: X+#|XX+* I can't find any documentation on it and it doesn't seem to match up with the patterns in Asterisk. i.e. ; X - any digit from 0-9 ; Z - any digit from 1-9 ; N - any digit from 2-9 ; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9) ; . - wildcard, matches anything remaining (e.g. _9011. matches ; anything starting with 9011 excluding 9011 itself) ; ! - wildcard, causes the matching process to complete as soon as ; it can unambiguously determine that no other matches are possible So what do + and | and * do? + means 'always add the part before the + if the part behind it matches'. ie: 0031+79NXX means if the number matches 79NXX (for instance 793456789) add 0031 (ie 00317934567890, which would be int'l format for the Netherlands) | means remove the part before if both the parts before and behind match. ie: 0031|79NXX means if the number matches 003179NXX (for instance 0031793456789) remove 0031 (ie 7934567890, which would be the national part of an int'l format number for the Netherlands) * means * (look on your phone's keypad G) (PS: + and | are only valid in certain contexts, such as trunks and outbound routes. Not everywhere | is valid, + will be valid as well, and vice versa! IIRC + only works for trunks) HTH -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] Flash operation on a call on a ZAP interface...
On Wed, December 7, 2005 2:24, Marc-andre Poupier said:
Also I have another question about the voicemail system, is it possible
in my message to say HI you have reached me blah blah if you need to
speak to so and so press on this number to reach him and the call would
be transferred back to an extension, like my extension 7 in this
example?
Not in VoiceMail, but you could make an autoattendant that does this...
(AMP has some easy features to help you do so)
An AMP Auto Attendant basically looks like this:
(Some options here rely on AMP macro's, so you may need to do things
slightly different in some cases!)
[aa_1]
;First put in the dial-options. Note that not all options need be
announced in the message:
exten = 0,1,Goto(aa_2,s,1) ;Go to another attendant (in my case,
switch languages)
exten = 1,1,Goto(ext-group,2,1);Dial group 2
exten = 2,1,Goto(ext-group,4,1);Dial group 4
exten = 3,1,Goto(ext-group,3,1);Dial group 3, etc.
exten = fax,1,Goto(ext-fax,in_fax,1) ;What to do if a fax called us
exten = h,1,Hangup() ;What to do when they hang up (hang up as well,
d'oh!)
exten = i,1,Playback(invalid) ;What to do on an invalid choice
exten = i,2,Goto(s,7) ;
include = ext-local ; include the local extension as valid options.
Allows direct dialling of extensions from the AA
include = app-messagecenter ; include the messagecenter. Allows direct
access to the message center from the AA
include = app-directory ; include the directory. Allows people to access
the 411 directory of asterisk
exten = s,1,GotoIf($[${DIALSTATUS} = ANSWER]?4);If the call was
already answered, go to #4 otherwise...
exten = s,2,Answer() ;Pickup
exten = s,3,Wait(1);Wait a bit before playing the AA message
exten = s,4,SetVar(DIR-CONTEXT=general);
exten = s,5,DigitTimeout(3); Set the timeout between digits
exten = s,6,ResponseTimeout(7) ; Set how long we'll wait on a choice
(timeout = invalid response, ie extension 'i', which sends you back to
's,7'
exten = s,7,Background(custom/aa_1) ; Play the AA message
(Note that anything behind a semi-colon (;) is a comment from myself to
explain what it is doing...)
HTH!
--
F Peeters
PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0
Cologne HFC-S pins #52, #54, #55 connected in parallel for synching.
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[Asterisk-Users] VoIP to GSM?
Hi all, I am looking for a cheap VoIP to GSM provider (most notably to GSM networks in The Netherlands), but so far the cheapest I have found is VoipGATE(.com, not .nl), and their prices are slightly (if using the Econo package) more expensive than the normal ISDN/PSTN rates The cheapest solution is still GSM-GSM, but SIP/IAX-GSM gateways are expensive gadgets, so I was wondering if anybody was aware of any provider that provides GSM termination at better rates than the Dutch KPN... http://www.kpn.com/kpn/show/id=879060/sc=7a7070 (I'm looking through the Wiki pages, but there's a lot of websites to check out when searching for rates... TIA! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] Sending data over ZAPHFC D-channel?
Is it possible to send data over the D Channel using ZAPHFC? I'd like to send data between three servers (only one is live yet, but I am thinking ahead and trying to plan...) to verify that each of their ISDN connections is live. Ie: 1 sends to 2 1 sends to 3 2 sends to 1 2 sends to 3 3 sends to 1 3 sends to 2 If this is possible, I could write an AGI script to notify on loss of ISDN link... TIA -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] ZapHFC cards not maintaining sync?!
On Fri, December 2, 2005 22:54, Francesco Peeters said: On Fri, December 2, 2005 22:50, Francesco Peeters said: On Fri, December 2, 2005 21:45, Kristof Hardy said: Francesco Peeters wrote: Does anybody have any experience in this? I am using * 1.2 BRIstuffed 0.3.0 Pre1 No experience on that, but there's an updated bristuff (0.3.0pre1b), maybe try that one? This is 1 issue that's fixed: - chan_zap/libpri fixes (stuck B channels) Just installed 0.3.0pre1c, but no change! :-/ I have now got this little ditty running to keep an eye on it: while true; do grep (F /proc/zaptel/2; sleep .1; done I do see the once a minute down-time come by as a combination of 1 F4, 2 F6's and then F7's. When it goes down for an extended time, it shows 1 F4 and a lot of F6's before finally returning F7's again... :-( Watching the console for a while I see regular messages, which I could also find in /var/log/messages: Dec 3 16:37:15 asterisk1 kernel: zaphfc[0]: received d channel frame with bad CRC. Dec 3 16:37:36 asterisk1 kernel: zaphfc[0]: empty HDLC frame received. Dec 3 16:37:36 asterisk1 kernel: zaphfc[0]: received d channel frame with bad CRC. Can anyone with one or more HFC-PCI card(s) (esp. in The Netherlands) check if they see these on a regular basis as well? (And I am talking many times an hour here!) TIA! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] ZapHFC cards not maintaining sync?!
On Sat, December 3, 2005 16:40, Francesco Peeters (Asterisk) said: On Fri, December 2, 2005 22:54, Francesco Peeters said: Watching the console for a while I see regular messages, which I could also find in /var/log/messages: Dec 3 16:37:15 asterisk1 kernel: zaphfc[0]: received d channel frame with bad CRC. Dec 3 16:37:36 asterisk1 kernel: zaphfc[0]: empty HDLC frame received. Dec 3 16:37:36 asterisk1 kernel: zaphfc[0]: received d channel frame with bad CRC. Can anyone with one or more HFC-PCI card(s) (esp. in The Netherlands) check if they see these on a regular basis as well? (And I am talking many times an hour here!) TIA! Ok, I have been analyzing the activities and see the following: 1) Every 10 seconds () the D channel gets torn down, which 2) Results in the CRC error, which means that 3) Every 3 minutes, the D channel goes down for EXACTLY 1 minute. This means there is a 66% chance of actually being able to use the ISDN link, and thus use it to dial out or be dialed on... This is obviously not acceptable for a PBX... I could try to get the KPN to give me a permanent D channel, but are there any tricks to try that would/could make asterisk somehow keep up the D channel?... -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] ZapHFC cards not maintaining sync?!
On Sat, December 3, 2005 19:01, Remco Barende said: On Sat, 3 Dec 2005, Francesco Peeters (Asterisk) wrote: On Sat, December 3, 2005 16:40, Francesco Peeters (Asterisk) said: On Fri, December 2, 2005 22:54, Francesco Peeters said: Watching the console for a while I see regular messages, which I could also find in /var/log/messages: Dec 3 16:37:15 asterisk1 kernel: zaphfc[0]: received d channel frame with bad CRC. Dec 3 16:37:36 asterisk1 kernel: zaphfc[0]: empty HDLC frame received. Dec 3 16:37:36 asterisk1 kernel: zaphfc[0]: received d channel frame with bad CRC. This is not normal. Run the florz patch over your bristuff install (I'm assuming you are using bristuff). These problems will cause your box to hang after anything beteen 5 and 48 hours. Already HAVE Florz patch installed! :-( What version of * and BRIstuff are you using? Can anyone with one or more HFC-PCI card(s) (esp. in The Netherlands) check if they see these on a regular basis as well? (And I am talking many times an hour here!) I am in NL :) I assumed as much when I saw your last name... :-) Whereabouts in NL? I'm in Zoetermeer (ZH)... 1) Every 10 seconds () the D channel gets torn down, which That's too slow, it should happen about every 1-2 seconds or so. The d channel going down and up again is normal behaviour. I know it is. Used to work for a Networking Competence Centre, and we had the same kind of issues with 3Com Netbuilders. The first call attempt after the D Channel was torn down always failed... The only solution was to get KPN to turn on the D Channel permanently... 2) Results in the CRC error, which means that 3) Every 3 minutes, the D channel goes down for EXACTLY 1 minute. I could try to get the KPN to give me a permanent D channel, but are there any tricks to try that would/could make asterisk somehow keep up the D channel?... I noticed that the 'deactivated' issue doesn't happen for a while after a call has been placed. I am now testing placing a call every minute, with a 100 ms timeout using the manager api. This means it never actually gets a chance to get through, or be picked up, but it does cause activity on the D channel. This has been running for half an hour now, and I haven't seen the channel go down for extended periods since. I'm not sure whether the KPN will like it, but it's an interesting test to run! G Good luck with our Royal Dutch KPN, but I would try florz first :) Tell me about it! Like I said above, we had *extensive* experience with them over the D Channel issue! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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
