Re: [Asterisk-Users] Voip / AudioCodes MP-108 Help Needed
Hi,Here are the step by step instructions for setting up a brand new AudiocodesFXS gateway for use with an Asterisk server: Connect the gateway to a network switch and connect a computer to the same switch. Then configure the IP address of the computer to 10.1.10.2. Then runyour web browser and point it to http://10.1.10.10 and login using the information below.Default IP address: 10.1.10.10Default user name: AdminDefault password: AdminGoto Quick Setup and change the following:IP Address = Set to the new IP address of the AudioCodes gateway Subnet Mask = Set to the correct netmask for your local network Default Gateway Address = Set to the correct gateway IP address for yourlocal network Working With Proxy = Set to Yes Proxy IP Address = Set to the IP address of the Asterisk server Enable Registration = Set to EnableRestart the gateway then log back in using the new IP address.Goto Protocol Management - Protocol Definition - Proxy Registration Registrar IP Address = Set to the IP address of the Asterisk server Registration Time = Set to 60Subscription Mode = Set to Per EndpointAuthentication Mode = Set to Per Endpoint Goto Protocol Management - Protocol Definition - DTMF Dialing Max Digits In Phone Num = Set to a large enough number such as 32Goto Protocol Management - Protocol Definition - Coders Add coders as needed You need to set at least G.711U-lawGoto Protocol Management - Endpoint Settings - Authentication Set SIP username and password for each portGoto Protocol Management - Endpoint Phone Numbers Enter an extension (phone) number for every used channelYour AudioCodes gateway is now ready.../Arun[EMAIL PROTECTED] www.intelegentnetworks.comOn 6/27/06, Mark Adams [EMAIL PROTECTED] wrote: Hello, Anyone here have experience with Audiocodes MediaPack MP-108 Gateways? I would be willing to pay someone for advice and support with configuring my gateways for a telemarketing project I am starting. My experience is somewhat limited but all I want to do is make outbound calls just like I would on normal pots lines. (That's the best way to explain it) I do not need any special configuration nor am I going to use it for any incoming calls. I would like to just have the gateways register and properly send calls out and relay DTMF tones. After I get them up and running I should have the manual read and digested by then and I will be good to go. Anyone interested please email me off list Mark Adams Infinity Marketing 216-334-9304 ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Changing standard Voicemail behavior
I am using Trixbox 1.0(Asterisk 1.2.7.1)at a customer site. They whishes to change the default Voicemail behavior. Standard behavior No answer/Busy - send to Voicemail Requested behavior No answer/Busy - message that if you press 9 you will instead be cent to reception - send to Voicemail or Reception if 9 pressed. I want this to always happen when Voicemail is invoked. How do I, inthe easiest way, change the default Voicemail behavior? Regards, Jan Berggren ---Jan Berggren Tel. 0371-83990intellIT AB Fax. 0371-83991S Storgatan 20 Mobil. 070-6210100SE-332 33 GISLAVED E-post arbete [EMAIL PROTECTED]Sverige Web. http://www.intellit.seSamarbetar med http://www.responsibility.se--- ___ --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] Why I can´t upload my wav file?
On Jun 27, 2006, at 12:19 PM, Yrving Rivas wrote: Hi everybody: Can somebody give me a hint? I have tried with gsm files, with wav 8khz 16bits, wav 8khz 8bits...and no way...what could be happening? Try asking a question that makes some sense? What are you trying to do? ___ --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] Addon-ooh323 install problem
On Jun 27, 2006, at 8:24 PM, Richard Scobie wrote: Tetsuya Yamamoto wrote: I can't makel asterisk addon, asterisk-ooh323. I use Asterisk and addons svn version. The current svn version of asterisk has had the module loader code redesigned and to date, the svn addons have not been updated to match this change. You will need to use the latest production versions of both. I wonder what this means? When you say Production version what are you referring to? Do you just mean the tar balls of 1.2.9 and latest addon? ___ --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] zaptel.conf settings for Singtel ISDN-2
Hi, Has anyone successfully configured a HFC ISDN card with Singtel's ISDN-2 service? If so, can you share the settings required? Thanks in advance. KokMeng. ___ --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] Most stable Asterisk version
On Jun 27, 2006, at 6:08 PM, shadowym wrote: Hi there, I am getting ready to set up a production Asterisk system. It needs to be stable. Upgrading, patching, rebooting, troubleshooting etc. are pretty much NOT an option once this thing is deployed. Like any phone system, it is expected to just work. Having said that, which is the best version and subversion of Asterisk to use? I was leaning towards 1.2 but it appears there are some major issues with it. At least with the most recent versions. Scary things like memory leaks and spontaneous crashing if the wrong and not exactly uncommon combination of events occur or the wrong and somewhat common features are used enough. I see that some of the commercial distributions are using 1.0.9 so I am thinking maybe that one. It has had some revisions to something like 1.0.11.1 or something like that so that is a tough call what to do there. I think there were some updates since 1.0 with BLF registration which is one thing I really would like to have working so another complicating factor I suppose. Anyone have any suggestions? What are some of the versions people are using that have impressive uptimes and just work? I don't think I need any of the new wiz bang stuff in 1.2. I just want something that works! I don't want to have to resort to scheduled nightly reboots either unless I don't have any other choice. 14 days, some hours, some minutes... I realize this isn't too impressive, but it's a pretty new version (1.2.9.1) ;~) It's been solid here. In general I tend to think that most of the issues you see here on the list are caused by mis-configurations. Asterisk definitely gives you enough rope to hang yourself and your neighbors extended family. Then again I am a newb and a lightweight also. If you are serious about your stated goal I suggest you build something and pound the snot out of it and see how it goes. I think things are going in a great direction and I look forward to 1.4! Also, it's not very hard to try to upgrade and then downgrade again if need be. Personally I would be more worried about hardware (as far as reliability goes). Marty PS I do NEED my asterisk to work at this point, and it does. ___ --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] Most stable Asterisk version
shadowym a écrit : Hi there, I am getting ready to set up a production Asterisk system. It needs to be stable. Upgrading, patching, rebooting, troubleshooting etc. are pretty much NOT an option once this thing is deployed. Like any phone system, it is expected to just work. Try FreeBSD's Asterisk port. It has been working rock-solid for me so far. It's been a few weeks now with no issues (fingers crossed)... But I admit that it does _JUST_ softswitching (i.e. call routing, load balancing and database CDR collection) and hence has the smallest possible feature set (search google: voip-info asterisk slimming). Another option is to buy Digium's commercial edition of Asterisk, which is supposed to be just what you describe. Best Regards, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE ___ --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] Help Asterisk crashes
Since no know had an answer me I finally figure this out. It was a corrupt sound file which was the source of this error. But I still don't understand why asterisk crashes when you have a corrupt sound file! Regards Fredrik Jensen -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fredrik Emil Jensen Sent: 27. juni 2006 12:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Help Asterisk crashes I am getting thousand of these messages in asterisk console Jun 27 12:35:55 WARNING[16496]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data And after some time the system crashes. Does anyone know why? I running Asterisk SVN-trunk-r7522 built Does it help to upgrade the system? Regards, Fredrik Jensen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Most stable Asterisk version
Same for me here, freebsd ports and same usage. Running from months on a Dell 1850(biXeon 4Gb ram) with no problems. Olivier Jean-Michel Hiver a écrit : shadowym a écrit : Hi there, I am getting ready to set up a production Asterisk system. It needs to be stable. Upgrading, patching, rebooting, troubleshooting etc. are pretty much NOT an option once this thing is deployed. Like any phone system, it is expected to just work. Try FreeBSD's Asterisk port. It has been working rock-solid for me so far. It's been a few weeks now with no issues (fingers crossed)... But I admit that it does _JUST_ softswitching (i.e. call routing, load balancing and database CDR collection) and hence has the smallest possible feature set (search google: voip-info asterisk slimming). Another option is to buy Digium's commercial edition of Asterisk, which is supposed to be just what you describe. Best Regards, Jean-Michel. ___ --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!!!
Ok, on peut parler français alors ;) Olivier Jean-Michel Hiver a écrit : Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten! ;-) Pues my punto fue que un poquito de correo en otro idioma no hace daño, y si ayuda mucho y molesta poco, ¿por qué quejarse? Quel bordel, sacrebleu! ___ --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] Addon-ooh323 install problem
Martin Joseph wrote: Do you just mean the tar balls of 1.2.9 and latest addon? Yes. I believe the svn addons package will be updated soon. ___ --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] can Asterisk act as a H.323 Gatekeeper?
Hallo. I managed to configure asterisk to act as H.323 gateway using asterisk built in support for H.323. I found it in ./channels/h323 directory of asterisk sources. I wonder whether asterisk can play a role of H.323 gatekeeper. If Yes, could You tell me some hints on how to do that. Greetings. ___ --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!!!
How many channels have you guys been able to get with this? The only problem I have with this is that it takes skype and a soundcard (virtual or otherwise) and the API is really executing commands on a running skype process. In my opinion its not worth it for 1 concurrent call per account. I have written code that works with skype in linux that simulates a virtual sound device. I have used that and successfully done calls out with this. I havent played with the dbus stuff (how you control the skype app from within linux) but since I have a soundcard that I know the audio format of it wouldnt be difficult to integrate this into asterisk, I could tweak chan_oss and make it into chan_skype fairly easily since that takes care of the other half of the equation. The only thing missing would be the events via dbus, which there are plenty of examples on so its not like all new code would have to be written. But its just not worth it if you have to have skype running for each call. And then you would potentially have to have a new username for each running process, and skype really wants X on linux so you would have to at least have the X virtual frame buffer (it works and acts like X but never displays anything or uses any hardware). That seems like an aweful lot of wasted resources on a box to connect to skype. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --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!!!
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 --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com wrote: How many channels have you guys been able to get with this? The only problem I have with this is that it takes skype and a soundcard (virtual or otherwise) and the API is really executing commands on a running skype process. In my opinion its not worth it for 1 concurrent call per account. I have written code that works with skype in linux that simulates a virtual sound device. I have used that and successfully done calls out with this. I havent played with the dbus stuff (how you control the skype app from within linux) but since I have a soundcard that I know the audio format of it wouldnt be difficult to integrate this into asterisk, I could tweak chan_oss and make it into chan_skype fairly easily since that takes care of the other half of the equation. The only thing missing would be the events via dbus, which there are plenty of examples on so its not like all new code would have to be written. But its just not worth it if you have to have skype running for each call. And then you would potentially have to have a new username for each running process, and skype really wants X on linux so you would have to at least have the X virtual frame buffer (it works and acts like X but never displays anything or uses any hardware). That seems like an aweful lot of wasted resources on a box to connect to skype. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461 DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can Asterisk act as a H.323 Gatekeeper?
Pawel wrote: I wonder whether asterisk can play a role of H.323 gatekeeper Not today. Although, disclaimed patches are gladly accepted at http://bugs.digium.com. Jeremy McNamara ___ --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] Avaya 4610sw SIP setup problem
Hi Tom, Thank you for your interest in my problem, I really am desperate about this thing... I have tried several versions one after another, and now I'm using the one released on 04.07.2006 (SIP release 2.2.2). Thanks, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom LynnSent: 28 June 2006 05:35To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem Which version of firmware are you using? On 6/27/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi all, I've been pulling my hair out for two days over this problem I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem is that the SIP PROXY, SIP DOMAIN and SIP REGISTRAR are simply ignored. The phone does take into account other values (WEB PROXY, etc), but it keps displaying "Registering" for ever. When I check the IP adresses, the SIP Proxy and Registrar fields are empty. This is not a network problem, I have made traces using Ethereal and I can see the right .txt file being transferred. Other settings in the file are applied too, just the SIP proxy and registrar are empty I have tried specifying them with and without quotes, by hostname, by IP address, Nada. It is all the more frustrating that everybody seems to have it working easily! Please help. Here is the contents of my 46xxsettings.txt file : SET DOMAIN mycompany.com SET DNSSRVR 204.140.111.43 SET PHNCC 352 SET PHNDPLENGTH 4 SET PHNIC 00 SET PHNOL 0 SET SYSLANG English SET APPSTAT 1 SET RESTORESTAT 1 SET AGCHAND 0 SET AGCHEAD 0 SET AGCSPKR 0 SET SNTPSRVR "204.140.111.200" SET DSTOFFSET "1" SET DSTSTART "1SunApr2L" SET DSTSTOP "LSunOct2L" SET GMTOFFSET "-5:00" SET DATESEPARATOR "/" SET DATETIMEFORMAT "3" SET DIALPLAN "[234]xxx|55" SET DIALWAIT "3" SET MUSICSRVR "" SET MWISRVR "" SET PHNNUMOFSA "3" SET REGISTERWAIT 120 SET SIPDOMAIN "sip.mycompany.com" SET SIPPROXYSRVR "204.140.111.219" SET SIPPORT "5070" (this is not a typo) SET SIPREGISTRAR "204.140.111.219" SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o=avaya .com IF $MODEL4 SEQ 4602 goto SETTINGS4602 IF $MODEL4 SEQ 4610 goto SETTINGS4610 IF $MODEL4 SEQ 4620 goto SETTINGS4620 IF $MODEL4 SEQ 4621 goto SETTINGS4621 IF $MODEL4 SEQ 4622 goto SETTINGS4622 IF $MODEL4 SEQ 4625 goto SETTINGS4625 IF $MODEL4 SEQ 4630 goto SETTINGS4630 goto END goto END SET WMLHOME http://support.avaya.com/elmodocs2/avayaip/4620/home.wml SET WMLPROXY 204.140.111.249 SET WMLPORT 3128 goto END goto END goto END goto END goto END SET WEBHOME http://support.avaya.com/elmodocs2/avayaip/4630/index.htm SET PHNEMERGNUM 112 goto END ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Avaya 4610sw SIP setup problem
Hi Tom, Thank you for your interest in my problem, I really am desperate about this thing... I have tried several versions one after another, and now I'm using the one released on 04.07.2006 (SIP release 2.2.2). Thanks, Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom LynnSent: 28 June 2006 05:35To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Avaya 4610sw SIP setup problem Which version of firmware are you using? On 6/27/06, Herchi Silviu [EMAIL PROTECTED] wrote: Hi all, I've been pulling my hair out for two days over this problem I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem is that the SIP PROXY, SIP DOMAIN and SIP REGISTRAR are simply ignored. The phone does take into account other values (WEB PROXY, etc), but it keps displaying "Registering" for ever. When I check the IP adresses, the SIP Proxy and Registrar fields are empty. This is not a network problem, I have made traces using Ethereal and I can see the right .txt file being transferred. Other settings in the file are applied too, just the SIP proxy and registrar are empty I have tried specifying them with and without quotes, by hostname, by IP address, Nada. It is all the more frustrating that everybody seems to have it working easily! Please help. Here is the contents of my 46xxsettings.txt file : SET DOMAIN mycompany.com SET DNSSRVR 204.140.111.43 SET PHNCC 352 SET PHNDPLENGTH 4 SET PHNIC 00 SET PHNOL 0 SET SYSLANG English SET APPSTAT 1 SET RESTORESTAT 1 SET AGCHAND 0 SET AGCHEAD 0 SET AGCSPKR 0 SET SNTPSRVR "204.140.111.200" SET DSTOFFSET "1" SET DSTSTART "1SunApr2L" SET DSTSTOP "LSunOct2L" SET GMTOFFSET "-5:00" SET DATESEPARATOR "/" SET DATETIMEFORMAT "3" SET DIALPLAN "[234]xxx|55" SET DIALWAIT "3" SET MUSICSRVR "" SET MWISRVR "" SET PHNNUMOFSA "3" SET REGISTERWAIT 120 SET SIPDOMAIN "sip.mycompany.com" SET SIPPROXYSRVR "204.140.111.219" SET SIPPORT "5070" (this is not a typo) SET SIPREGISTRAR "204.140.111.219" SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o=avaya .com IF $MODEL4 SEQ 4602 goto SETTINGS4602 IF $MODEL4 SEQ 4610 goto SETTINGS4610 IF $MODEL4 SEQ 4620 goto SETTINGS4620 IF $MODEL4 SEQ 4621 goto SETTINGS4621 IF $MODEL4 SEQ 4622 goto SETTINGS4622 IF $MODEL4 SEQ 4625 goto SETTINGS4625 IF $MODEL4 SEQ 4630 goto SETTINGS4630 goto END goto END SET WMLHOME http://support.avaya.com/elmodocs2/avayaip/4620/home.wml SET WMLPROXY 204.140.111.249 SET WMLPORT 3128 goto END goto END goto END goto END goto END SET WEBHOME http://support.avaya.com/elmodocs2/avayaip/4630/index.htm SET PHNEMERGNUM 112 goto END ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 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] Problem with callerid in sip to isdn gateway
On 6/27/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
On Tue, 2006-06-27 at 15:00 +0200, Morten Isaksen wrote: Hi! I have this setup:
PABX --ISDN30-- Asterisk 1 --SIP-- Asterisk 2 --ISDN30-- TELCO Digium TE410P is used in both Asterisk 1 and 2. When I set the CLIR bit on the PABX the Callerid / ANI is removed
somewhere between the SIP interface on Asterisk 1 and the SIP interface on Asterisk 2.Have you used a packet sniffer to ensure that its actually sent toasterisk 2?If it isnt then that may be the entire problem.Before
trying to diagnose anything on the isdn side I would make sure that itis infact being sent correctly.Alternatively you can try some noops()on asterisk2 for when a call is received to display the caller id to the
console, that may be easier for some than reading sip headers.
On Asterisk 1 the ${CALLERID(num)} is correct but on Asterisk 2 CALLERID(num) is set to Unknown if CALLINGPRES=32. If CALLINGPRES=0 then the CALLERID(num) is passed to Asterisk 2.
I have solved the problem this way:
On Asterisk 1:
exten = _[2-9]XXX,1,sipaddheader(x-clir: ${CALLINGPRES})exten = _[2-9]XXX,n,setcallerpres(allowed_not_screened)exten = _[2-9]XXX,n,dial(SIP/[EMAIL PROTECTED]
)
On Asterisk 2:
exten = _X.,n,set(CLIR=${SIP_HEADER(x-clir)})exten = _X.,n,gotoif($[$[${CLIR}=32]]?NOCID:CID)exten = _X.,n(NOCID),SetCallerPres(prohib_not_screened)exten = _X.,n(CID),dial(ZAP/g2/${EXTEN})
-- Morten Isaksenhttp://www.misak.dk/blog/
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Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!
On Wed, 2006-06-28 at 08:14 +, [EMAIL PROTECTED] wrote: 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. it wouldnt need a real soundcard, that is part of the point. I remap all the calls the same way that I did for allowing instant porting of your digium g729 licenses (in another post, code is at my personal site http://www.0xdecafbad.com/ somewhere). Remapping those calls is trivial, there are very few things that are acutally done to a soundcard to set it up, ioctl() for setting the sample rate, etc and read/write/open/close basically. Really trivial code. It would however be nicer if you didnt have to run a seperate copy of the binary for each call. This has a direct cost against memory. It would be better if it didnt use memory to open a GUI (even with the virtual framebuffer for X it still takes all that memory even though it doesnt display for real). I also doubt that a 386 would cut it, with everything going on it would have to be faster and that pushes the cost up. If you are going to do that it might be cheaper to buy one of the 1,2,4 port FSX/FXO devices for integrating with a phone system or something (some plug into wall jacks others into phones). The 4 ports are about $750 which is steep. The 99 port one which is unclear how you use it exactly is $1500 or so. Actualy looking at the 99 port model it appears that its just a usb soundcard that has a FXS port on it, which is a silly way in my opinion, and still requires a system running skype to work :( 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 I dont, the overhead is insane. As as for a price for 'them' it would just be a software program. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --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.conf settings for Singtel ISDN-2
KokMeng Loh wrote: Hi, Has anyone successfully configured a HFC ISDN card with Singtel's ISDN-2 service? If so, can you share the settings required? The AVM Fritz! PCI works fine with chan_capi for Singtel. At that time (about 2 years back), I didn't test HFC because the driver was very immature. What kind of problems do you have? Did you try to connect through a TA box (the NT-1) or direct? Cheers. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems compiling Zaptel
Hi guys, I'm getting the following error when trying to compile zaptel on a debian machine running 2.4.27-3-386. gcc -g -c -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o zttool.o zttool.c gcc -lnewt zttool.o -o zttool gcc -shared -Wl,-soname,libtonezone.so.1.0 -lm -o libtonezone.so zonedata.lo tonezone.lo make[1]: Leaving directory `/usr/src/zaptel' mkdir -p /dev/zap rm -f /dev/zap/ctl rm -f /dev/zap/channel rm -f /dev/zap/pseudo rm -f /dev/zap/timer rm -f /dev/zap/253 rm -f /dev/zap/252 rm -f /dev/zap/251 rm -f /dev/zap/250 mknod /dev/zap/ctl c 196 0 mknod /dev/zap/transcode c 196 250 mknod: `/dev/zap/transcode': File exists make: *** [devices] Error 1 And when I try to modprobe the X100P card, I get the following. asterisk:/usr/src/zaptel# modprobe wcfxo /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol remove_wait_queue_R5dbd8645 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol __pollwait_R43c77cc3 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol create_proc_entry_Ra52db232 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol proc_mkdir_Rba727c62 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol add_wait_queue_Rf89d8ae0 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol remove_proc_entry_Rf2afedc2 /lib/modules/2.4.27-3-386/misc/zaptel.o: insmod /lib/modules/2.4.27-3-386/misc/zaptel.o failed /lib/modules/2.4.27-3-386/misc/zaptel.o: insmod wcfxo failed Any help is much appreciated. Regards, Mark. ___ --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] getting agentID and DNID help
Hi Guys
I have just installed a call center onto Suse 10. I have managed to do a
DBget (astdb) and extract the DNID numbers to play a DNID specific
greeting. We have installed Snom 320 and the customer would like us to
Send the DNID(nam) to the phone screens so that the agent will be able
to answer in the correct language and with the specific customer company
name (ie. Agent says Welcome to JP Morgan helpline as they will be
able to get the JP morgan helpline name on the screen). This is what my
incoming dial-plan looks like so far:
[Inbound}
exten = _X.,1,GotoIfTime(21:29-07:29|mon-thu|*|*?After-hours,s,1)
exten = _X.,2,GotoIfTime(17:59-07:59|fri-sat|*|*?After-hours,s,1)
exten = _X.,3,GotoIfTime(17:59-07:29|sun|*|*?After-hours,s,1)
exten = _X.,4,Answer
exten = _X.,5,Wait(2)
exten = _X.,6,NoOp(-ID ${CALLERID(num)}-)
exten = _X.,7,DBget(COMPDNID=checked/${DNID})
exten = _X.,8,GotoIf($[${DNID} = 4966]?clientX,4966,1:9)
exten = _X.,9,GotoIf($[${COMPDNID} = 1]?10:102)
exten = _X.,10,ResponseTimeout(7)
exten = _X.,11,Background(custom/${DNID}1)
exten = 1,1,Goto(english-q,1,1)
exten = 2,1,Goto(lang-2-q,1,1)
exten = 3,1,Goto ...etc
I have tried to do a:
exten = _X.,1,Set(CALLERID(name)=${DB(CIDNAMES/${DNID})})
but this has not seemed to help.
I am sure there is someone out there that will be able to point me in
the right direction. Please bear in mind that i am still fairly new to
asterisk and their dial plans.
The second question i would like to ask is as follows:
I have roaming agents. They have no fixed seating positions as the work
in shifts for 24 hour days. There are extensions on the desks and they
have unique agent login ID's.
This creats a small problem if i am trying to pause the agents from the
multiple queues that are in. 5 of them to be correct.
This is what i was thinking:
exten = PauseQueueMember(|Agent/${agentid}) ;
but i cant seem to manage to get the agentID from anywhere, I see that
once an agent has logged in there is a global variable
AGENTBYCALLERID_{EXTEN} set. Is there some way of using this to pause or
unpause the agent?
TIA
Terry
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Re: [Asterisk-Users] getting agentID and DNID help
Hi,
On 11:31, Wed 28 Jun 06, Terry Wade wrote:
Hi Guys
[snip]
I have tried to do a:
exten = _X.,1,Set(CALLERID(name)=${DB(CIDNAMES/${DNID})})
but this has not seemed to help.
Use this: exten = _X.,1,LookupCIDName()
That works great in my setup
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer afficionados are both called users?
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[Asterisk-Users] password on radius authentication
Hi, It's kind of off-topic , but still within Asterisk. I developed an asterisk module that send an authentication to a radius server for call authorization and process its reply (limited to User-Name and Cisco or Quintum VSA h323 attribute). My question, is when it make sense to use or include the attribute Password/User-Password? Looking on PDF's of Quintum and Cisco none of it really make use of this attribute. Any comment? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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, Jun 28, 2006 at 08:14:56AM -, [EMAIL PROTECTED] wrote: 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 You would be violating the terms of usage of their API if you want to use (let alone sell) such a device. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems compiling Zaptel
On Wed, Jun 28, 2006 at 05:22:17PM +0800, Mark Davies wrote: Hi guys, I'm getting the following error when trying to compile zaptel on a debian machine running 2.4.27-3-386. What kernel-headers/kernel-source? I hope you didn't extract a source tarball of 2.4.27/debian and linked it to /usr/src/linux . apt-get install kernel-headers-`uname -r` gcc -g -c -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o zttool.o zttool.c gcc -lnewt zttool.o -o zttool gcc -shared -Wl,-soname,libtonezone.so.1.0 -lm -o libtonezone.so zonedata.lo tonezone.lo make[1]: Leaving directory `/usr/src/zaptel' mkdir -p /dev/zap rm -f /dev/zap/ctl rm -f /dev/zap/channel rm -f /dev/zap/pseudo rm -f /dev/zap/timer rm -f /dev/zap/253 rm -f /dev/zap/252 rm -f /dev/zap/251 rm -f /dev/zap/250 mknod /dev/zap/ctl c 196 0 mknod /dev/zap/transcode c 196 250 mknod: `/dev/zap/transcode': File exists make: *** [devices] Error 1 And when I try to modprobe the X100P card, I get the following. asterisk:/usr/src/zaptel# modprobe wcfxo /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol remove_wait_queue_R5dbd8645 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol __pollwait_R43c77cc3 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol create_proc_entry_Ra52db232 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol proc_mkdir_Rba727c62 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol add_wait_queue_Rf89d8ae0 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol remove_proc_entry_Rf2afedc2 /lib/modules/2.4.27-3-386/misc/zaptel.o: insmod /lib/modules/2.4.27-3-386/misc/zaptel.o failed /lib/modules/2.4.27-3-386/misc/zaptel.o: insmod wcfxo failed This reminds me of the error messages that required the change 2.4.27-1 = 2,4,27-2 . -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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, 2006-06-28 at 13:15 +0300, Tzafrir Cohen wrote: 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 You would be violating the terms of usage of their API if you want to use (let alone sell) such a device. I am unsure if all the hardware devices are basically usb soundcards or not, havent really looked, but if they arent then it would seem to me that its possible to do. Further I dont think it would be against their api to write sofeware that uses their api. That is what was being discussed when this comment came out, so ... -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --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] HDLC Bad FCS (8)
Hi All. Somebody of you already passed below for this error? Jun 28 02:25:03 NOTICE[31148 ]: chan_zap.c:8207 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Jun 28 02:52:08 NOTICE[31148 ]: chan_zap.c:8207 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Jun 28 04:11:10 NOTICE[31148 ]: chan_zap.c:8207 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Jun 28 04:29:29 NOTICE[31148 ]: chan_zap.c:8207 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span1 I am not detectingfails in link, have one asterisk-1.2.9.1 linked with a central office Siemens HiPath 4000 and I believe it is functioning, although the times the call to be completed without Ring, nor audio. I hug to all Best Regards Josué ___ --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] Work required - modify Asterisk + SEMS
Hi all, I am looking for a developer or developers that can implement the following: - Modify an Asterisk server in order to support one inbound RTP and several outbound RTPs, I was thinking SEMS may provide a very good starting point. The idea is to make a PA system over IP. We do *not* want full-duplex audio. - Implement a client in Qt/C++, that allows to send audio to this platform, and plays back audio received from it (Windows-based). We are thinking about Speex for the codec, as there are no royalty issues. Interested parties please reply with your comments, capabilities, so we can start discussing the project. Best regards, Mike ___ --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] a2billing
I am using a2billing as billing system on tixbox but I have a problem since the user call the destination number ,the ivr tell him about him amount and ask him to enter the destination number ,my question is how can I let the user call the destination directly . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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!!!
Hi, Is it illegal to use Uplink Skype2Sip software to connect a skype account to a homepbx asterisk? ( Just to know... i don't want to be bored because of asteriskpt.blogspot) On 6/28/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Wed, 2006-06-28 at 13:15 +0300, Tzafrir Cohen wrote: 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 You would be violating the terms of usage of their API if you want to use (let alone sell) such a device. I am unsure if all the hardware devices are basically usb soundcards or not, havent really looked, but if they arent then it would seem to me that its possible to do. Further I dont think it would be against their api to write sofeware that uses their api. That is what was being discussed when this comment came out, so ... -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (GNU/Linux) iD8DBQBEol9n+1olxlzQw5cRAvmYAJ463UBN/3F1bkCo3smt92QaQhPzOACfSn/j OijC0wHuU8hmynUp/Osa6gA= =hEQW -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Com os melhores cumprimentos, Marco Mouta ___ --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] In other words (Why I can´t u pload my wav file?)
I am trying to upload a wav file to my asterisk through the AMP. That file is going to be used as the receptionist voice. The amp requires the file to be sampled to 8khz 16 bit which is done. I am using the Freepbx portal to upload the file. That file comes from a record I have paid for to a professional narrator. I would like help in this matter, given that I am stuck at this point since 5 days. Help, please. Yrving [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Miércoles, 28 de Junio de 2006 02:27 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Why I can´t upload my wav file? On Jun 27, 2006, at 12:19 PM, Yrving Rivas wrote: Hi everybody: Can somebody give me a hint? I have tried with gsm files, with wav 8khz 16bits, wav 8khz 8bits...and no way...what could be happening? Try asking a question that makes some sense? What are you trying to do? ___ --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 __ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - http://correo.yahoo.com.mx/ ___ --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 auto-dial Help
Hi,When you originate a call asterisk essentially callouts to the Specified channel and the when answers connects the the context,extension,priority. What if I want my dial plan to make the origination call and the destination call. What would I specify for my dialplan/callout file?thanks in advance../Arun ___ --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] a2billing
Hi,you check your a2billing.conf file:; Please enter here the file you want to play when we prompt the calling party to enter his destination number; file_conf_enter_destination = prepaid-enter-number-u-calling-1-or-011 file_conf_enter_destination = prepaid-enter-destI think this file should help you.../Arun[EMAIL PROTECTED] www.intelegentnetworks.comOn 6/28/06, Khaled Chehab [EMAIL PROTECTED] wrote: I am using a2billing as billing system on tixbox but I have a problem since the user call the destination number ,the ivr tell him about him amount and ask him to enter the destination number ,my question is how can I let the user call the destination directly . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO for PSTN
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Or a TDM2400 with 4 FXO modules... (4x4=16) :) Lito Lampitoc wrote: oh sorry, 2 TDM400P with 4 FXO modules each :=) On 6/28/06, *Lito Lampitoc* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: or TDM400P with four FXO modules perhaps? On 28 Jun 2006 01:47:05 -, [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: 1 FXO per PSTN, so you would need 16 FXO ports. That would be accomplished by 4 TDM100P with 4 FXO modules on each. Undrhil --- Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com wrote: If I have 16 PSTN for my trunklines, how many FXO do I need? Thanks. Lito ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEonCP1Kolm8VQlAURAnukAJ44TuB1yhu3Msu4ubMqi5gyDkZbbQCgvaG9 YdMuNeI+y0evoNFIkkBFcGk= =4jvJ -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HDLC Bad FCS (8)
Hi, Take a look here: http://www.asteriskguru.com/tutorials/hdlc_bad_fcs.htmlit might help. Otherwise you can also try different settings for the "span" line in zaptel.conf Silviu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josué ContiSent: 28 June 2006 12:33To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] HDLC Bad FCS (8) Hi All. Somebody of you already passed below for this error? Jun 28 02:25:03 NOTICE[31148 ]: chan_zap.c:8207 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Jun 28 02:52:08 NOTICE[31148 ]: chan_zap.c:8207 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Jun 28 04:11:10 NOTICE[31148 ]: chan_zap.c:8207 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 Jun 28 04:29:29 NOTICE[31148 ]: chan_zap.c:8207 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span1 I am not detectingfails in link, have one asterisk-1.2.9.1 linked with a central office Siemens HiPath 4000 and I believe it is functioning, although the times the call to be completed without Ring, nor audio. I hug to all Best Regards Josué ___ --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] Trixbox maunual configuration
I love the added apps installed with trixbox, ARI, Web-Meetme, FOP, and Reports are great. FreePBX on the other hand, is nearly impossible to do everything with. Trying to edit the configs manually proves impossible due to the excessive use of includes and macros. It is kind of like watching someone try to bite their own ear off. Has anybody tried to wipe all the configs clean and program the switch manually. Will this interfere with the other apps. I would wipe out extensions.conf, voicemail.conf, IAX.conf SIP.conf queues.conf and agents.conf. I do not want to use the FreePBX again after this. I am not trying to put down FreePBX, I know a lot of people have worked very hard on this. It just over complicates things for me. Jordan Novak Communications Technician ___ --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] Voicemail volume adjustment
This works great, however, when I look at the full log, it says that the sendmail is executing prior to vm-audio. Any way to change this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cullin J. Wible Sent: Tuesday, June 27, 2006 8:41 PM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Voicemail volume adjustment In voicemail.conf: externnotify=/opt/asterisk-1.2.7.1/sbin/vm-audio The attached script should increase as much as possible without clipping. Cheers, Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Technical Support Sent: Tuesday, June 27, 2006 3:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail volume adjustment I frequently find voice messages are emailed to users with insufficient volume - barely audible. I would like to have asterisk run a sox command to adjust the volume of each message before emailing (perhaps once the message has been left). Has anyone done this? Care to share the steps? Thanks, MD ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] point to point T hookup?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA256 I have an installation where I'll have a site to site data DS1 for use between two corporate offices. We'll have one asterisk server at each office. I'd like to be able to route calls over the 24 channels on that DS1 between the offices, instead of over the voiceT at each location to maximize savings on interoffice calls. An alternative is to put a router and switch at each end and extend a data network to the other site for SIP traffic. Would that result in better quality calls? What configuration areas are there to be set and how are they diffent than just a standard PRI, which I have working now? Thanks for your help, Jonathan -BEGIN PGP SIGNATURE- Version: PGP Universal 2.0.6 iQEVAwUBRKJtFpJhYmFK+jfsAQh/tggAiqCqlefhEyAuIcshX5AaMGx3flVdHn5C mh1TY5i/Z8tf4LBEh+TuXvUFGNXvnPn12nrEwkF8s4HOUcDwVhAXI5XlA7WZFT83 H3UGoK7RGaitirWHDKFEfa3+BlWpL8eclsdItGx0FPHtdQeRCxq2ba1gtKszpaHC KgApM9ExYVwEPFcwbYoK2m0pvofuiYNYxw/yN7ZkIooM1oWTP8NFjGuysrb2FW2J 8odHb+J8ySmhmHQFWZ+XVHnkOTckp+feaKUuCohsffBxBm5mPrdXpQMwnCCR5yhz bhoAaveMPJz7gcSIgXTAMyZtO4m8U3/zht443S1J/MTD30seL8goPg== =r6vs -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trixbox maunual configuration
On 07:23, Wed 28 Jun 06, Jordan Novak wrote: I love the added apps installed with trixbox, ARI, Web-Meetme, FOP, and Reports are great. FreePBX on the other hand, is nearly impossible to do everything with. Trying to edit the configs manually proves impossible due to the excessive use of includes and macros. It is kind of like watching someone try to bite their own ear off. Has anybody tried to wipe all the configs clean and program the switch manually. Will this interfere with the other apps. I would wipe out extensions.conf, voicemail.conf, IAX.conf SIP.conf queues.conf and agents.conf. I do not want to use the FreePBX again after this. I am not trying to put down FreePBX, I know a lot of people have worked very hard on this. It just over complicates things for me. Hi, They are all seperate opensource packages. None of them depend on eachother. We use FOP, ARI and Reports (areski stats) here too without FreePBX. Simply download them and look at their README/INSTALL files. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Two FXO: How to dial a number when a RING comes in?
Hi Vincent - Sorry for the long delay in responding. I didn't see you message until now due to the postfix problems on the mailing list. Anyway, I see some clues here: exten = s,1,Answer exten = s,2,Waitexten(10) exten = 100,Dial(Zap/2/014XX) Then call in and after you're connected, dial 100 to see if it will dial out on ZAP/2 When I try this, /var/log/asterisk/messages says: Jun 19 18:12:38 NOTICE[1660] pbx.c: Cannot find extension '100' in context '(null)' I know you mentioned that you forgot the '100', but more importantly, the log says you are in context '(null)', which is not good. Make sure you have a context in extensions.conf, and make sure your fxo cards are pointed to go to it in zapata.conf. For reference, I went back to the original configuration that I used, but it picks up the line and remains silent (static noises): I think Eric Wieling is right. You have another problem not related to what you are trying to do in the dialplan. It sounds like one of your fxo cards or one of your phone lines is not working properly (or maybe both). Test both phone lines and both interfaces by dialing into both of them (make sure they are pointed to a context in the extensions.conf, and make sure they have something to do there when you try to dial). Can you get in to the asterisk box at all? Then try swapping the phone lines with the fxo interfaces. Can you dial in then? If you need help writing some testing configs like this, just let me know. - Noah ___ --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] FXO X100P
Hi, I have a 100P card but even though there's no incoming route, it answers the line after 2 to 3 rings. If I do create an incoming route, the same happens, but it never rings the ring group or extension I enter. It's almost as if the card acts as a modem. The caller hears nothing, just silence. I have a VoIP incoming route which works perfect. Can anyone assist me? Many thanks, Pierre ___ --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] Trixbox maunual configuration
I can confirm this. AMP/TrixBox is a wonderful project but if you like to tweak something or you became a moreexperienced user, it will became soonas a straitjacket. I'm still struggling to clean AMP config files to work with a plain Asterisk install. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jordan NovakSent: Wednesday, June 28, 2006 2:24 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Trixbox maunual configuration I love the added apps installed with trixbox, ARI, Web-Meetme, FOP, and Reports are great. FreePBX on the other hand, is nearly impossible to do everything with. Trying to edit the configs manually proves impossible due to the excessive use of includes and macros. It is kind of like watching someone try to bite their own ear off. Has anybody tried to wipe all the configs clean and program the switch manually. Will this interfere with the other apps. I would wipe out extensions.conf, voicemail.conf, IAX.conf SIP.conf queues.conf and agents.conf. I do not want to use the FreePBX again after this. I am not trying to put down FreePBX, I know a lot of people have worked very hard on this. It just over complicates things for me. Jordan Novak Communications Technician ___ --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] point to point T hookup?
On Wednesday 28 June 2006 08:48, Jonathan Miller wrote: An alternative is to put a router and switch at each end and extend a data network to the other site for SIP traffic. Would that result in better quality calls? If you can ensure that voice traffic has top priority in all the routers between the two sites, there should be no difference in voice quality. For a true point-to-point system this is trivial to achieve, and maximizes the bang-for-buck ratio of your interoffice connection. Obviously having two ADSL connections is not true point to point -- you will want a leased line, or a dedicated connection to a common provider who has the prioritization of voice traffic in your SLA. You could, in theory, have higher than telco quality voice calls with a VOIP system, as you are no longer restricted to 8kHz-sampled, 16-bit audio. Naturally the phones must support this for this to work. What configuration areas are there to be set and how are they diffent than just a standard PRI, which I have working now? If you put a point-to-point DS1 between sites, it's easy. Asterisk can act as a PRI CPE or CO endpoint. -A. ___ --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] F3000 registering to asterisk
Neil Cherry wrote: [snip] How did you get access to the web config? What user and is it the default password/access code? type it's IP address into a web browser. Username: admin, password: psw is the default. cheers, Paul. ___ --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] h323 phone
I installed an asterisk server with oh323 channel driver support. Then I uploaded the H323 firmware on a AT320 phone (Usually I use it as a sip phone, but I am using it just for test) Let's say that I assigned 945 as phone number, account and password to this phone, and its ip address were 192.168.1.88 Which are the right entries to add in /etc/asterisk/oh323.conf ? I tried (with no chance..) [945] type=user username=945 secret=945 host=192.168.1.88 context=from-internal incominglimit=4 thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --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] point to point T hookup?
Hi Jonathan - I have an installation where I'll have a site to site data DS1 for use between two corporate offices. We'll have one asterisk server at each office. I'd like to be able to route calls over the 24 channels on that DS1 between the offices, instead of over the voiceT at each location to maximize savings on interoffice calls. An alternative is to put a router and switch at each end and extend a data network to the other site for SIP traffic. Would that result in better quality calls? You'll get better quality calls by using the 24 channels of the T1 directly as voice channels. They'll be high-quality ulaw calls, but you'll be limited to 23 simultaneous calls over the link. If that's OK with the client, I'd go that route. You can avoid QoS setup and jitterbuffer configuration. On the other hand, if they want more simultaneous calls than that over this link, you could use it as a data T1, and use g729, and you could fit a LOT more calls over this link. They'll be lower quality just because of the g729 codec, and you also have to deal with QoS and jitterbuffer. - Noah ___ --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] point to point T hookup?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA256 I have a true leased line (a T1) between the two sites. What parts do I configure for Asterisk to utilized the link bi-directional? On Wednesday 28 June 2006 09:09, Andrew Kohlsmith wrote: On Wednesday 28 June 2006 08:48, Jonathan Miller wrote: An alternative is to put a router and switch at each end and extend a data network to the other site for SIP traffic. Would that result in better quality calls? If you can ensure that voice traffic has top priority in all the routers between the two sites, there should be no difference in voice quality. For a true point-to-point system this is trivial to achieve, and maximizes the bang-for-buck ratio of your interoffice connection. Obviously having two ADSL connections is not true point to point -- you will want a leased line, or a dedicated connection to a common provider who has the prioritization of voice traffic in your SLA. You could, in theory, have higher than telco quality voice calls with a VOIP system, as you are no longer restricted to 8kHz-sampled, 16-bit audio. Naturally the phones must support this for this to work. What configuration areas are there to be set and how are they diffent than just a standard PRI, which I have working now? If you put a point-to-point DS1 between sites, it's easy. Asterisk can act as a PRI CPE or CO endpoint. -A. ___ --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 -BEGIN PGP SIGNATURE- Version: PGP Universal 2.0.6 iQEVAwUBRKJ3ZZJhYmFK+jfsAQjdewf/YoFUn6lO3EFp9qDBxDQTS+aVRrdyqNfp b2K4SGyk2iC70/3h0jPZBNHtcjuM7YTtLFniuy9JgOjbM6mniGBNitCeOD6x1o0D gLg5WJ5BlprFNQFHZX03ItaaJ/PLax0W2VJr7h0YxIyFZ19euO90W569PXeWJLJk UkrnhB4HkWV7VqS9YDAMl0MWcNcmNpuMxnf+sv1Csbf8muGj9t/4WraZ5Ac0K7l6 drSIMJtfTlFPe6TPrB7A5B9nZgxMtwRHq8gPr6ki3NnNGw2O/dscuFaVRMfVg6U0 UmNVPBd9ViUakwPudSmujzhePG6GPYI3Zb6U3uRUO/9qRQZjeCsPiw== =Jgu7 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] isdn-data over iax
is the following zaptel.conf configuration correct for TDMoE used for pri-cpe signalling - is this possible at all ? I couldn't find an example... loadzone=nl defaultzone=nl # pri E1 card span=1,1,3,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 # hfc-pci 1 span=2,1,3,ccs,ami bchan=32-33 dchan=34 # hfc-pci 2 span=3,1,3,ccs,ami bchan=35-36 dchan=37 #TDMoE dynamic=eth,eth0/00:D0:09:E8:FA:EB,31,0 bchan=38-52 dchan=53 bchan=54-68 [EMAIL PROTECTED] schrieb am 27.06.2006 17:01:26: [EMAIL PROTECTED] wrote: is it possible to route an ISDN-Data channel over an iax-connection ? the setup is pc with isdn-card - (zaphfc) Asterisk Server1 (iax) - (iax) Asterisk Server2 (E1)-connecting to an external isdn-dialin router via the iax-line the call is transfered as speech which is not accepted at the remote end IAX is not suited for this. Maybe TDMoE is an option for you ? Florian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.7.1 with Polycom 501 on SIP - Conf Calling
Do you have more than one call per line enabled on the Poly? Is it the phone or asterisk returning the busy? What does the console say? On Jun 27, 2006, at 5:29 PM, Mike Staver wrote: I have one extension setup for each Polycom 501 I have, and when I try to call out on a conference call, I get all circuits busy for the second call. I have one sip trunk set up for each DID that I have through our VoIP provider. Each trunk is capable of having one call placed on it at one time. So, I'm thinking I need a way to tell Asterisk to have the second call go out on one of the other empty trunks at the time if one exists, which more than likely, it will. Is this possible? -- -Mike Staver [EMAIL PROTECTED] [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] point to point T hookup?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 What kind of T1? TDM? Data? What type of signaling are you planning to use em? There is a lot of information that that question is lacking for anyone to advise you ... Jonathan Miller wrote: I have a true leased line (a T1) between the two sites. What parts do I configure for Asterisk to utilized the link bi-directional? On Wednesday 28 June 2006 09:09, Andrew Kohlsmith wrote: On Wednesday 28 June 2006 08:48, Jonathan Miller wrote: An alternative is to put a router and switch at each end and extend a data network to the other site for SIP traffic. Would that result in better quality calls? If you can ensure that voice traffic has top priority in all the routers between the two sites, there should be no difference in voice quality. For a true point-to-point system this is trivial to achieve, and maximizes the bang-for-buck ratio of your interoffice connection. Obviously having two ADSL connections is not true point to point -- you will want a leased line, or a dedicated connection to a common provider who has the prioritization of voice traffic in your SLA. You could, in theory, have higher than telco quality voice calls with a VOIP system, as you are no longer restricted to 8kHz-sampled, 16-bit audio. Naturally the phones must support this for this to work. What configuration areas are there to be set and how are they diffent than just a standard PRI, which I have working now? If you put a point-to-point DS1 between sites, it's easy. Asterisk can act as a PRI CPE or CO endpoint. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEoo9t1Kolm8VQlAURAhA8AKCRLcbtqAwyWL/auoDwB/hB87tjfQCfTly2 21GPSotFeMZQduIw51c99P8= =w2r1 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] point to point T hookup?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA256 Your response leads me to further question this setup... It's a full data T that is not provisioned. Being that I control the termination at each end, do I get to specify the encoding? On Wednesday 28 June 2006 10:17, Sean Cook wrote: What kind of T1? TDM? Data? What type of signaling are you planning to use em? There is a lot of information that that question is lacking for anyone to advise you ... Jonathan Miller wrote: I have a true leased line (a T1) between the two sites. What parts do I configure for Asterisk to utilized the link bi-directional? On Wednesday 28 June 2006 09:09, Andrew Kohlsmith wrote: On Wednesday 28 June 2006 08:48, Jonathan Miller wrote: An alternative is to put a router and switch at each end and extend a data network to the other site for SIP traffic. Would that result in better quality calls? If you can ensure that voice traffic has top priority in all the routers between the two sites, there should be no difference in voice quality. For a true point-to-point system this is trivial to achieve, and maximizes the bang-for-buck ratio of your interoffice connection. Obviously having two ADSL connections is not true point to point -- you will want a leased line, or a dedicated connection to a common provider who has the prioritization of voice traffic in your SLA. You could, in theory, have higher than telco quality voice calls with a VOIP system, as you are no longer restricted to 8kHz-sampled, 16-bit audio. Naturally the phones must support this for this to work. What configuration areas are there to be set and how are they diffent than just a standard PRI, which I have working now? If you put a point-to-point DS1 between sites, it's easy. Asterisk can act as a PRI CPE or CO endpoint. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: PGP Universal 2.0.6 iQEVAwUBRKKE+JJhYmFK+jfsAQgltAf/U8jI+M953rGssIPngoWR+QBXS8NYt59q SufUjwhxGJ8Vd1tlnFS4t4OzkJ7csD3Nfz65LEH7fW7/kS6ac2U/WOu5s53+imc+ Ter+0qK2sSwnSzP4eW364TFY4aDH7cxaHa8iqcjuUl3YnowMni29YMu/pa1fD7eD Rfykz8Pb5pzYx/ojEp0akT9HXW44xUV65+1bakfJJPPDv4sMfLrw69KQGnsHd42t ZqzxZYlp9GU76ice4dvwMOcRI5KZbDbXqkEx+r+ZA39E3Acap2rjDg4sAZxKa+8h LxBOoOhugn7TjOJKUva4L48HslkhO9bTJuQc7Iqdu3CoiNUTHMgw+A== =YPlN -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] point to point T hookup?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Typically with a data t1 you are running either HDLC or PPP on either end. I assume you have a cisco router on either end? Or are you planning to plug asterisk with a Digium/Sangoma/Other T1 card? Personally if it is a data t1 I would use a cisco router then do QoS on both routers and do everything VoIP on the asterisk side... Then you have no hardware necessary for your trunks (other than the routers of course) Jonathan Miller wrote: Your response leads me to further question this setup... It's a full data T that is not provisioned. Being that I control the termination at each end, do I get to specify the encoding? On Wednesday 28 June 2006 10:17, Sean Cook wrote: What kind of T1? TDM? Data? What type of signaling are you planning to use em? There is a lot of information that that question is lacking for anyone to advise you ... Jonathan Miller wrote: I have a true leased line (a T1) between the two sites. What parts do I configure for Asterisk to utilized the link bi-directional? On Wednesday 28 June 2006 09:09, Andrew Kohlsmith wrote: On Wednesday 28 June 2006 08:48, Jonathan Miller wrote: An alternative is to put a router and switch at each end and extend a data network to the other site for SIP traffic. Would that result in better quality calls? If you can ensure that voice traffic has top priority in all the routers between the two sites, there should be no difference in voice quality. For a true point-to-point system this is trivial to achieve, and maximizes the bang-for-buck ratio of your interoffice connection. Obviously having two ADSL connections is not true point to point -- you will want a leased line, or a dedicated connection to a common provider who has the prioritization of voice traffic in your SLA. You could, in theory, have higher than telco quality voice calls with a VOIP system, as you are no longer restricted to 8kHz-sampled, 16-bit audio. Naturally the phones must support this for this to work. What configuration areas are there to be set and how are they diffent than just a standard PRI, which I have working now? If you put a point-to-point DS1 between sites, it's easy. Asterisk can act as a PRI CPE or CO endpoint. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEopTt1Kolm8VQlAURAjfOAKCeMdAejVE1HLtdXr8xMO5G0lLlVACgo87v sLfeVfN+mvLp1ovNuo1BBVg= =wOGP -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] JAMAICA DID'S - 1-876
Oliver Vermeulen wrote: JAMAICA DID'S - 1-876 NOW ACTIVE ON www.didx.org Oliver Vermeulen World Venture Group Telecom Corporate Address: 147 New Haven Point Lane West Palm Beach , FL , Miami USA DID: +1 (305)722-1457 BE DID: +(32)9-395-5620 UK DID: +(44)870-478-8896 SIP : [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] msn: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.wvg-tele.com ___ --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 Please stop using this list for email which belongs on the -biz list. You already sent it over there. That should be enough. W ___ --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] point to point T hookup?
Assuming it is a dedicated private line p2p T1 Assuming that 23 calls at one time is sufficient Install a T1 card in each server, plug the T1 in and set one end ofr pri net, the other for pri cpe. zaptel.conf and zapata.conf are the files you are looking for. Just define the 23 channels as a group and dial by the group number. Using pri will pass callerid info for you across the connection On Jun 28, 2006, at 9:30 AM, Jonathan Miller wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA256 Your response leads me to further question this setup... It's a full data T that is not provisioned. Being that I control the termination at each end, do I get to specify the encoding? On Wednesday 28 June 2006 10:17, Sean Cook wrote: What kind of T1? TDM? Data? What type of signaling are you planning to use em? There is a lot of information that that question is lacking for anyone to advise you ... Jonathan Miller wrote: I have a true leased line (a T1) between the two sites. What parts do I configure for Asterisk to utilized the link bi-directional? On Wednesday 28 June 2006 09:09, Andrew Kohlsmith wrote: On Wednesday 28 June 2006 08:48, Jonathan Miller wrote: An alternative is to put a router and switch at each end and extend a data network to the other site for SIP traffic. Would that result in better quality calls? If you can ensure that voice traffic has top priority in all the routers between the two sites, there should be no difference in voice quality. For a true point-to-point system this is trivial to achieve, and maximizes the bang-for-buck ratio of your interoffice connection. Obviously having two ADSL connections is not true point to point -- you will want a leased line, or a dedicated connection to a common provider who has the prioritization of voice traffic in your SLA. You could, in theory, have higher than telco quality voice calls with a VOIP system, as you are no longer restricted to 8kHz-sampled, 16-bit audio. Naturally the phones must support this for this to work. What configuration areas are there to be set and how are they diffent than just a standard PRI, which I have working now? If you put a point-to-point DS1 between sites, it's easy. Asterisk can act as a PRI CPE or CO endpoint. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: PGP Universal 2.0.6 iQEVAwUBRKKE+JJhYmFK+jfsAQgltAf/U8jI+M953rGssIPngoWR+QBXS8NYt59q SufUjwhxGJ8Vd1tlnFS4t4OzkJ7csD3Nfz65LEH7fW7/kS6ac2U/WOu5s53+imc+ Ter+0qK2sSwnSzP4eW364TFY4aDH7cxaHa8iqcjuUl3YnowMni29YMu/pa1fD7eD Rfykz8Pb5pzYx/ojEp0akT9HXW44xUV65+1bakfJJPPDv4sMfLrw69KQGnsHd42t ZqzxZYlp9GU76ice4dvwMOcRI5KZbDbXqkEx+r+ZA39E3Acap2rjDg4sAZxKa+8h LxBOoOhugn7TjOJKUva4L48HslkhO9bTJuQc7Iqdu3CoiNUTHMgw+A== =YPlN -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] point to point T hookup?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 It always helps to read the original post... so I apologize. I think what you are looking to do is route the calls over the existing data t1 in which case all you need to do is create an IAX trunk between the two asterisk servers addressing their internal ip addresses ( such that the route would be over the data T1). Then you would want to make sure that you are running QoS on all voip traffic that goes from A to B accross that link giving it the highest level of priority. That should in essence do the job... Sean -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEopYe1Kolm8VQlAURAu6RAKDQq6JqNPI/JNwufyKtOXfKeL6vkwCgsIjE k2iGKkNL1IXwvxtcNbxZJbo= =Q5K3 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] JAMAICA DID'S - 1-876
JAMAICA DID'S - 1-876 NOW ACTIVE ON www.didx.org Oliver Vermeulen World Venture Group Telecom Corporate Address: 147 New Haven Point Lane West Palm Beach , FL , Miami USA DID: +1 (305)722-1457 BE DID: +(32)9-395-5620 UK DID: +(44)870-478-8896 SIP : [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] msn: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.wvg-tele.com ___ --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] Getting at SIP error with SIP_HEADER() ?
Hi, when attempting to dial an invalid number with Nikotel this is returned: SIP/2.0 400 Bad Request request uri is neighter my realm or a valid dns and Asterisk prints smth similar on the CLI. However it appears that I cannot get access to 400 Bad Request from the dialplan because this error is not part of any SIP header, and therefore the function SIP_HEADER won't do the trick. Right or wrong? ;-) Cheers, Philipp ___ --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.conf settings for Singtel ISDN-2
Hi Leo, How stupid of me! I just realized that I needed a NT-1 box between my HFC card and the ISDN line! What I observed was that the line was always not active. Thanks for your reply anyway. -kokmeng. Leo Ann Boon wrote: KokMeng Loh wrote: Hi, Has anyone successfully configured a HFC ISDN card with Singtel's ISDN-2 service? If so, can you share the settings required? The AVM Fritz! PCI works fine with chan_capi for Singtel. At that time (about 2 years back), I didn't test HFC because the driver was very immature. What kind of problems do you have? Did you try to connect through a TA box (the NT-1) or direct? Cheers. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial Tone + EM
Maybe one of you can help me with this: We have T1's that come from both MCI and Global Crossing as uses channelized (24 Ports per T) with inband (DTMF) ANI and DNIS delivery (format = *DNIS*ANI*). My old equipment was set for D4, AMI, SF and Wink Start and so is Asterisk Server. I've moved these T's to Asterisk TE410P and inbound calls are arriving to external voice mail correctly (Dialogic D240-SC-T1) - without issues. I guess you recognize these are NOT PRI T1's - but old style DS1. However, when the external voice mail system begins to dial out, it grabs the port waits for the Wink and expects dial tone to be returned afterwards - Hearing none, it just sits there until the time out and gives up. My thinking is there should be an EM signaling type that CAN provide dial tone. - A quick scan of the source (chan_zap.c), it appears there is no such provisions for DT for any of the EM types. To me it appears to be a simple patch, but I'm sure I would screw it up if I attempt this myself, not being a programmer. And if by chance I would get it working, the next update would also need that patch. I'm hoping I can find someone on the list that is willing to add a new EM method with a DT provision and make it available to the release sources Thanks Bart = Zaptel.conf # Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 AMI/D4 RED span=1,0,0,d4,ami em=1-24; = seems like my only choice (em) # Span 2: TE4/0/2 TE410P (PCI) Card 0 Span 2 AMI/D4 RED span=2,0,0,d4,ami em=25-48 ; = seems like my only choice (em) Zapata.conf: ; Span 1: TE4/0/1 TE410P (PCI) Card 0 Span 1 ; This is attached to CUST 3 VMS System ; signalling =em_w ; = might be wrong choice (see below for others) context=default group = 1 channel = 1-24 ; Span 2: TE4/0/3 TE410P (PCI) Card 0 Span 3 ; This T1 is WorldCom Local 714 DID's ; signalling =em_w ; = might be wrong choice (see below for others) context=from-did group = 3 channel = 25-48 Anybody have a clue for me TIA Bart ___ --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] Re: Changing standard Voicemail behavior
Hi, We have this set up in a few places. Basically, what you need to do is play a sound file using the Background() command. This allows for user entry. The sound file will say press 9 for reception, or stay on the line to leave a voicemail. This will go in the no answer/busy priority of your dialplan. I could give you coding examples if you'd like, but I'm not familiar with Trixbox, and if there is something different about it than a regular Asterisk system. Email me if you'd like more help. Leah Newmark Capalon VoIP [EMAIL PROTECTED] wrote: Message: 1 Date: Wed, 28 Jun 2006 08:18:53 +0200 From: Jan Berggren [EMAIL PROTECTED] Subject: [Asterisk-Users] Changing standard Voicemail behavior To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii I am using Trixbox 1.0(Asterisk 1.2.7.1)at a customer site. They whishes to change the default Voicemail behavior. Standard behavior No answer/Busy - send to Voicemail Requested behavior No answer/Busy - message that if you press 9 you will instead be cent to reception - send to Voicemail or Reception if 9 pressed. I want this to always happen when Voicemail is invoked. How do I, in the easiest way, change the default Voicemail behavior? Regards, Jan Berggren --- Jan Berggren Tel. 0371-83990 intellIT AB Fax. 0371-83991 S Storgatan 20 Mobil. 070-6210100 SE-332 33 GISLAVED E-post arbete [EMAIL PROTECTED] Sverige Web. http://www.intellit.se/ http://www.intellit.se http://www.intellit.sesamarbetar/ Samarbetar med http://www.responsibility.se http://www.responsibility.se/ --- ___ --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] Mysql Trixbox
Hello, I have installed FreeRadius server on Trixbox Server. My problem is mysql is not letting FreeRadius to login either locally or remotely. I also insert proper entries in HOST and USERS tables. But it does not work I always get ERROR 1045 (28000); Access Denied for user 'root'@'localhost' Thanks Wazb ___ --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 shutdown
Guys. Ive seen on my asterisk messages log that asterisk has shutdown itself about 12 times in 5 days... The logs show nothing but: [Jun 28 09:40:02] WARNING[3172]: Unicall/4 event Drop call [Jun 28 09:40:02] WARNING[3172]: Unicall/4 event Release call [Jun 28 09:40:02] VERBOSE[3172]: [Jun 28 09:40:02] -- Unicall/4 released [Jun 28 09:40:02] VERBOSE[3084]: [Jun 28 09:40:02] Asterisk cleanly ending (15). [Jun 28 09:40:03] VERBOSE[28320]: [Jun 28 09:40:03] Asterisk Event Logger Started /var/log/asterisk/event_log [Jun 28 09:41:01] VERBOSE[28368]: [Jun 28 09:41:01] Beginning asterisk shutdown [Jun 28 09:41:01] VERBOSE[28368]: [Jun 28 09:41:01] Executing last minute cleanups [Jun 28 09:41:01] VERBOSE[28368]: [Jun 28 09:41:01] Asterisk cleanly ending (15). [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] Asterisk Event Logger Started /var/log/asterisk/event_log [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Parsing '/etc/asterisk/dnsmgr.conf': [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Parsin g '/etc/asterisk/dnsmgr.conf': [Jun 28 09:41:01] Found [Jun 28 09:41:01] NOTICE[28457]: Managed DNS entries will be refreshed every 1200 seconds. [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] Asterisk Dynamic Loader loading preload modules: [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Parsing '/etc/asterisk/modules.conf': [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Parsi ng '/etc/asterisk/modules.conf': [Jun 28 09:41:01] Found [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Manager registered action Ping [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Manager registered action Events [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Manager registered action Logoff [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Manager registered action Hangup [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Manager registered action Status [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Manager registered action Setvar [Jun 28 09:41:01] VERBOSE[28457]: [Jun 28 09:41:01] == Manager registered action Getvar As you can see, there are no noticeable errors or anything so.. Anybody has seen this before? Is there any way to make asterisk more verbose? Im running it as -cg Any hints? ___ --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] (no subject)
Hey everybody, Is it alright to run two TDM400s on the same machine? If it is, how would one differentiate between the channels on each card? So, if Im running strait FXS and my first card is fxsks 1-4, would the second be fxsks 5-8? Would there be any interrupt problems? Any help would be great! Thanks! Tj ___ --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 1.2.8 compilation problem
Hi all I have downloaded asterisk 1.2.8 try to make on RHEL AS 4 i get the following error any clue make[1]: Entering directory `/root/all/asterisk-1.2.8/res'make[1]: Nothing to be done for `all'.make[1]: Leaving directory `/root/all/asterisk-1.2.8/res'make[1]: Entering directory `/root/all/asterisk-1.2.8 /channels'gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -fomit-frame-pointer -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -fPIC -c -o chan_sip.o chan_sip.c chan_sip.c: In function `sip_send_mwi_to_peer':chan_sip.c:11358: warning: unused variable `s'chan_sip.c: In function `sip_poke_peer':chan_sip.c:11620: error: `s' undeclared (first use in this function) chan_sip.c:11620: error: (Each undeclared identifier is reported only oncechan_sip.c:11620: error: for each function it appears in.)make[1]: *** [chan_sip.o] Error 1make[1]: Leaving directory `/root/all/asterisk- 1.2.8/channels'make: *** [subdirs] Error 1[EMAIL PROTECTED] asterisk-1.2.8]# ___ --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] (no subject)
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
The only issues you could potentially run into is if all the modules
are FXS and they all needed to ring simultaneously... your power
supply may not be suited to handle to voltage requirements.
Sean
Ninneman, Tj wrote:
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Hey everybody,
Is it alright to run two TDM400s on the same machine? If it is,
how would one differentiate between the channels on each card? So,
if I?m running strait FXS and my first card is fxsks 1-4, would the
second be fxsks 5-8? Would there be any interrupt problems?
Any help would be great!
Thanks!
Tj
--
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[Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.
Hello, Here is a breakdown of the issue I am experiencing. I have three remote employees, in various states, who have Polycom 501 phones. They are unable to receive incoming calls after a few minutes of the phones being plugged in. They work immediately after being plugged in, but they lose the ability shortly thereafter. They can always make outbound calls, but only to real phone numbers, not extensions. They each have NAT routers, and I have triple checked that they have opened/forwarded the correct ports, basically 5060-3 UDP. Once they plug the phone it (power and ethernet) I see on the CLI console of the asterisk server that the phones register: Asterisk CVS-v1-0-12/01/04-18:46:01, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk CVS-v1-0-12/01/04-18:46:01 currently running on bell (pid = 3652) nell*CLI Verbosity is at least 10 -- Registered SIP '3015' at XXX.XXX.XXX.XXX port 1500 expires 3600 Here is the top part of my sip.conf ;_ ;sip.conf ;_ [general] port=5060 bindaddr=0.0.0.0 externip=XXX.XXX.XXX.XXX localnet=XXX.XXX.XXX.XXX/255.255.255.248 canreinvite=no tos=reliability srvlookup=yes disallow=all allow=ulaw dtmfmode=rfc2833 nat=yes ignoreregexpire=yes I know it has something to do with the NAT because if I plug my Polycom directly into my cable modem, thus making it sit on the Internet and have a real IP, everything works just fine. I am curious what I am missing. Thanks. Von L. ___ --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] Voicemail volume adjustment
Good catch - I hadn't realized that. You are correct that in app_voicemail.c sendmail is run prior to the externnotify script. I see a few options: 1) change the code in app_voicemail.c 2) Use the externotify script to assemble and send the email messages 3) Run a web server and include a link to the voicemail message instead of attaching it. None of them look fun. Not sure how many * developers read this list, but it would be great of the run_externnotify(vmu-context, vmu-mailbox); call in notify_new_message() in app_voicemail.c could be moved to the top of the function as it is probably the preferred solution. Cullin J. Wible Co-Founder CTO Email Data Source, Inc. 212-514-8900 x1006 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins Sent: Wednesday, June 28, 2006 8:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Voicemail volume adjustment This works great, however, when I look at the full log, it says that the sendmail is executing prior to vm-audio. Any way to change this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cullin J. Wible Sent: Tuesday, June 27, 2006 8:41 PM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Voicemail volume adjustment In voicemail.conf: externnotify=/opt/asterisk-1.2.7.1/sbin/vm-audio The attached script should increase as much as possible without clipping. Cheers, Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Technical Support Sent: Tuesday, June 27, 2006 3:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail volume adjustment I frequently find voice messages are emailed to users with insufficient volume - barely audible. I would like to have asterisk run a sox command to adjust the volume of each message before emailing (perhaps once the message has been left). Has anyone done this? Care to share the steps? Thanks, MD ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mysql Trixbox
This isn't anything asterisk is causing. Sounds to me like FreeRadius is not properly authenticating with mysql. Example: If I wanted to log into a mysql box remotely, it could be down like this $ mysql -p -h XXX.XXX.XXX.XXX -u username databaseName Which means that 'username' better be in the users table. If you do something like this $mysql -h XXX.XXX.XXX.XXX databaseName You will prob get that error you are seeing. Its more detailed then this, but that might help. Von Wasif wrote: Hello, I have installed FreeRadius server on Trixbox Server. My problem is mysql is not letting FreeRadius to login either locally or remotely. I also insert proper entries in HOST and USERS tables. But it does not work I always get ERROR 1045 (28000); Access Denied for user 'root'@'localhost' Thanks Wazb ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.2.7.1 with Polycom 501 on SIP - ConfCalling
We run them with 1 call per line, but when we first set them up they would do 8. The problem was switching between calls on a single line. At that time, however, the phone did not return busy and allowed the calls to stack up. This is set in the XML configuration files. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Wednesday, June 28, 2006 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk 1.2.7.1 with Polycom 501 on SIP - ConfCalling Do you have more than one call per line enabled on the Poly? Is it the phone or asterisk returning the busy? What does the console say? On Jun 27, 2006, at 5:29 PM, Mike Staver wrote: I have one extension setup for each Polycom 501 I have, and when I try to call out on a conference call, I get all circuits busy for the second call. I have one sip trunk set up for each DID that I have through our VoIP provider. Each trunk is capable of having one call placed on it at one time. So, I'm thinking I need a way to tell Asterisk to have the second call go out on one of the other empty trunks at the time if one exists, which more than likely, it will. Is this possible? -- -Mike Staver [EMAIL PROTECTED] [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Trixbox maunual configuration
As pointed out, just build your own system. If you understand the Freepbx dialplan, you can usually do almost anything you want in _custom files including redefining contexts in such a way that upgrades do not wipe them out. It's simply a matter of spending some time to see what is being done and then extending it. On the other hand - there are some more difficult scenarios to get around and plenty of other good reasons to just role your own. You can have one, the other or the best of both if that has value to you and you are willing to understand the dialplan, config and how to integrate into and work with it. (I use all three, depending on the scenario)pFrom: "Mimmus" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.comDate: Wed, 28 Jun 2006 15:04:53 +0200Subject: RE: [Asterisk-Users] Trixbox maunual configuration I can confirm this. AMP/TrixBox is a wonderful project but if you like to tweak something or you became a moreexperienced user, it will became soonas a straitjacket. I'm still struggling to clean AMP config files to work with a plain Asterisk install. From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of JordanNovakSent: Wednesday, June 28, 2006 2:24 PMTo:asterisk-users@lists.digium.comSubject: [Asterisk-Users] Trixboxmaunual configuration I love the added apps installedwith trixbox, ARI, Web-Meetme, FOP, and Reports are great. FreePBX on theother hand, is nearly impossible to do everything with. Trying to edit theconfigs manually proves impossible due to the excessive use of includes andmacros. It is kind of like watching someone try to bite their own ear off. Hasanybody tried to wipe all the configs clean and program the switch manually.Will this interfere with the other apps. I would wipe out extensions.conf,voicemail.conf, IAX.conf SIP.conf queues.conf and agents.conf. I do not wantto use the FreePBX again after this. I am not trying to put down FreePBX, Iknow a lot of people have worked very hard on this. It just over complicatesthings for me. JordanNovak CommunicationsTechnician ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Next-gen email? Have it all with the all-new Yahoo! Mail Beta.___ --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] CONSOLE/dsp
how can i make exten = 780836,1,Dial(CONSOLE/dspSIP/,40,m) the CONSOLE (ALSA) not to accepts the call always? Juergen alsa.conf is autoanswer=no ___ --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] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.
You have to lower the registration interval in the phones to under a minute otherwise the NAT hole closes and no calls come in. Polycom has said that they are going to be putting in a keep alive in the firmware at some point. On 6/28/06, Von L. [EMAIL PROTECTED] wrote: Hello, Here is a breakdown of the issue I am experiencing. I have three remote employees, in various states, who have Polycom 501 phones. They are unable to receive incoming calls after a few minutes of the phones being plugged in. They work immediately after being plugged in, but they lose the ability shortly thereafter. They can always make outbound calls, but only to real phone numbers, not extensions. They each have NAT routers, and I have triple checked that they have opened/forwarded the correct ports, basically 5060-3 UDP. Once they plug the phone it (power and ethernet) I see on the CLI console of the asterisk server that the phones register: Asterisk CVS-v1-0-12/01/04-18:46:01, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk CVS-v1-0-12/01/04-18:46:01 currently running on bell (pid = 3652) nell*CLI Verbosity is at least 10 -- Registered SIP '3015' at XXX.XXX.XXX.XXX port 1500 expires 3600 Here is the top part of my sip.conf ;_ ;sip.conf ;_ [general] port=5060 bindaddr=0.0.0.0 externip=XXX.XXX.XXX.XXX localnet=XXX.XXX.XXX.XXX/255.255.255.248 canreinvite=no tos=reliability srvlookup=yes disallow=all allow=ulaw dtmfmode=rfc2833 nat=yes ignoreregexpire=yes I know it has something to do with the NAT because if I plug my Polycom directly into my cable modem, thus making it sit on the Internet and have a real IP, everything works just fine. I am curious what I am missing. Thanks. Von L. ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.
On 12:04, Wed 28 Jun 06, Von L. wrote: Hello, ;_ ;sip.conf ;_ [general] port=5060 bindaddr=0.0.0.0 externip=XXX.XXX.XXX.XXX localnet=XXX.XXX.XXX.XXX/255.255.255.248 canreinvite=no tos=reliability srvlookup=yes disallow=all allow=ulaw dtmfmode=rfc2833 nat=yes ignoreregexpire=yes Show us one of the phone entries. Basically check if the following is set there: nat=yes qualify=yes The qualify=yes will send packets so the nat states stay open. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk ACD with Polycom IP501
BJ, One other thing, did I need to have a version of asterisk already installed before your version? I had a blank system with Debian installed lipri-1.2.3 (make clean, make, make install) installed zaptel-1.2.6 (as above) done svn checkout http:..functions asterisk-polycom cd into asterisk-polycom did make clean, make, make install, make samples Edited the samples to get it to work. Does that sound right? Thanks again for you help, Dean. -Original Message- From: Dean @ INKnBITs [mailto:[EMAIL PROTECTED] Sent: 28 June 2006 11:22 To: BJ Weschke Subject: RE: [Asterisk-Users] Asterisk ACD with Polycom IP501 The show version has the following: Asterisk SVN-bweshke-polycom_acd_functions-r36151 built by I have done the sip trace, not sure if it makes a file to pull off, but the screen shows: ---(14 headers 0 lines)--- Creating new subscription sending to 192.1.3.103 :5060 (no NAT)- this is the correct IP for the phone Found peer '501' Looking for in demo (domain 192.1.3.101)- correct asterisk ip Transmitting (no NAT) to 192.1.3.103:5060: SIP/2.0 404 Not Found Hope that helps, if you need any more lines or if there is a file I can pull. Thanks, Dean. -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: 27 June 2006 12:39 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk ACD with Polycom IP501 On 6/27/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote: I'm new to this and don't know how to do a sip trace, but have attached the files as requested. Thanks for your help. Dean. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of BJ Weschke Sent: 26 June 2006 15:21 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk ACD with Polycom IP501 Hi Dean - It should be working. If not, please email me a sip debug trace along with your /etc/asterisk/agents.conf and your /etc/asterisk/sip.conf. Thanks. BJ On 6/26/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote: Hi, Has anybody got the polycom acd function to work? I have the following setup: Debian 3.1 - 2.6.8 linux zlib-1.1.4 libpri-1.2.3 zaptel- 1.2.6 Asterisk - the bweschke/polycom_acd_funtions branch version - I get one error when doing a make install about needing a newer version of libpri and zaptel, I got the above versions from asterisk.org, are there newer version anywhere else? In the sip.conf file I have set the agentlogin=yes and agentcbcontext=demo (demo as from extensions.conf context) I have setup an agent in agents.conf as ,1234,Name I have changed in the sip.cfg of the polycom phone: feature.15.name=acd-login-logout feature.15.enabled=1 feature.16.name=acd-agent-availability feature.16.enabled=1 and in the phone1.cfg of the polycom I'm only using line1 so made the changes below: reg.1.acd-login-logout=1 reg.1.acd-agent-available=1 I get the login button on the phone, and when I try and login with the agent it just goes back to login. Hi. We really need a sip debug to try and capture what's happening here. Enable/Uncomment the full line in your logger.conf file and then issue sip debug from your CLI and then try your agent login again. With that, we'll be able to see behind the scenes what's going on. Additionally, please tell me what you get when you do a show version from the CLI. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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] In other words (Why I can ´t upload my wav file?)
On Jun 28, 2006, at 4:45 AM, Yrving Rivas wrote: I am trying to upload a wav file to my asterisk through the AMP. That file is going to be used as the receptionist voice. The amp requires the file to be sampled to 8khz 16 bit which is done. I am using the Freepbx portal to upload the file. That file comes from a record I have paid for to a professional narrator. I would like help in this matter, given that I am stuck at this point since 5 days. I don't know anything about AMP or FreePBX, as I use plain old asterisk. I have found that for messages and prompts you want the recording to be in the codec format that is most used by your system (ie GSM, or for me uLaw). This prevents transcoding and saves CPU. WAV isn't actually a specific data type as far as I can see, it's more of a file type that umbrella's over several actual formats. You probably have your audio in the wrong format. Many people on this list use sox to convert audio formats. Audacity is another option. Using asterisk to record your audio is a certain way to get a proper format :~) Good Luck, Marty ___ --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] Remote employees using Polycom 501 lose
The Polycom's need to have their registration time lowered. Set it to 60 seconds which will re-register every 30 seconds. The polycom doesn't have any sort of 'keep alive' feature to keep the NAT holes open. There is information on the wiki fruther describing this and how to set it up if you don't know where to look.pFrom: "Von L." [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Wed, 28 Jun 2006 12:04:40 -0400Subject: [Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes. Hello,Here is a breakdown of the issue I am experiencing. I have three remoteemployees, in various states, who have Polycom 501 phones. They areunable to receive incoming calls after a few minutes of the phones beingplugged in. They work immediately after being plugged in, but they losethe ability shortly thereafter. They can always make outbound calls, butonly to real phone numbers, not extensions.They each have NAT routers, and I have triple checked that they haveopened/forwarded the correct ports, basically 5060-3 UDP. Once theyplug the phone it (power and ethernet) I see on the CLI console of theasterisk server that the phones register:Asterisk CVS-v1-0-12/01/04-18:46:01, Copyright (C) 1999-2004 Digium.Written by Mark Spencer <[EMAIL PROTECTED]>=Connected to Asterisk CVS-v1-0-12/01/04-18:46:01 currently running onbell (pid = 3652)nell*CLIVerbosity is at least 10-- Registered SIP '3015' at XXX.XXX.XXX.XXX port 1500 expires 3600Here is the top part of my sip.conf;_;sip.conf;_[general]port=5060bindaddr=0.0.0.0externip=XXX.XXX.XXX.XXXlocalnet=XXX.XXX.XXX.XXX/255.255.255.248canreinvite=notos=reliabilitysrvlookup=yesdisallow=allallow=ulawdtmfmode=rfc2833nat=yesignoreregexpire=yesI know it has something to do with the NAT because if I plug my Polycomdirectly into my cable modem, thus making it sit on the Internet andhave a real IP, everything works just fine.I am curious what I am missing.Thanks.Von L.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.___ --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] Remote employees using Polycom 501 lose ability to
Von L. wrote: Asterisk CVS-v1-0-12/01/04-18:46:01, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk CVS-v1-0-12/01/04-18:46:01 currently running on I would suggest you upgrade your Asterisk. This is VERY outdated and CVS to boot! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h263 Video Support Questions
Hi, What asterisk release (stable or dev) has support for a softphone like Xlite (free) that uses h263 for video codec? (audio works fine) in sip.conf I added [xlite1] videosupport=yes allow=h263 allow=gsm nat=yes canreinvite=no Also, what (proven/tested) hardphones with video support can be used with asterisk? Im using 1.2.7 stable release. thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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 shutdown
Anton Krall wrote: Guys. Ive seen on my asterisk messages log that asterisk has shutdown itself about 12 times in 5 days... The logs show nothing but: What version? I'm running 1.2.9.1 and saw one of my Asterisk process, this morning, just shut down for no apparent reason. I didn't have a console at the time. The logs don't show anything. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ztdummy and Debian on Intel Macmini
Hi,Is there a way to by-pass the absence of ztdummy on a Debian powered Intel Macmini platform ?Regards ___ --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] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.
FYI, when we had NAT routers at both locations setting qualify=yes did not work. On 6/28/06, Michiel van Baak [EMAIL PROTECTED] wrote: On 12:04, Wed 28 Jun 06, Von L. wrote: Hello, ;_ ;sip.conf ;_ [general] port=5060 bindaddr=0.0.0.0 externip=XXX.XXX.XXX.XXX localnet=XXX.XXX.XXX.XXX/255.255.255.248 canreinvite=no tos=reliability srvlookup=yes disallow=all allow=ulaw dtmfmode=rfc2833 nat=yes ignoreregexpire=yes Show us one of the phone entries. Basically check if the following is set there: nat=yes qualify=yes The qualify=yes will send packets so the nat states stay open. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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] WIFI sip phone
Hi folks!Based upon your experience on the field what wifi sip phone would youreccomend ?A customer asked for a wireless * install and I'm looking for advice, tnxAlessio Focardi[[*] - Interconnessioni Italy ___ --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] Voicemail volume adjustment
Why use an application like sox - when you can make the voicemail application do it natively: exten = s,1,Dial(SIP/100,10) exten = s,2,Voicemail([EMAIL PROTECTED]|ug(10)) The key is the g(10) parameter: From the 'show application voicemail': g(#) - Use the specified amount of gain when recording the voicemail message. The units are whole-number decibels (dB). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Technical Support Sent: Tuesday, June 27, 2006 3:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail volume adjustment I frequently find voice messages are emailed to users with insufficient volume - barely audible. I would like to have asterisk run a sox command to adjust the volume of each message before emailing (perhaps once the message has been left). Has anyone done this? Care to share the steps? Thanks, MD ___ --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] DUNDi Not Able to HandleComplexFailoverSituations
-Original Message- From: Michael Collins [mailto:[EMAIL PROTECTED] Sent: Monday, June 26, 2006 4:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] DUNDi Not Able to HandleComplexFailoverSituations I get annoyed Stephen when Digium goes around calling Asterisk 'enterprise grade', which in my opinion it really isn't. I'd consider distributed ACD queues to be a requirement for an enterprise grade product, but it's becoming apparent that there is no mechanism for implementing this. I'm being told that DUNDi isn't the right man for the job. I'd suggest you ask Digium for your money back. Leif. Question: isn't there a bigger picture issue here? I've seen a lot of bashing going on in this thread but not very much useful dialogue. (Doug bashing DUNDi and Digium, other people bashing Doug for his annoying posts, etc.) Whether or not we like or dislike Doug's tone is, IMHO, irrelevant. How about we tackle the REAL questions: If DUNDi isn't the answer, what is? Is Asterisk even capable of doing what Doug needs, namely, distributed ACD queues? If so, how? If not, why? Is it even feasible to try to do it? Will it require an Asterisk add-on, or can the core be modified to do this? (This leads to the question for the dev list...) These questions, of course, lead to other questions: If * can be programmed to do distributed ACD queues, does that mean there are other features that might benefit from a distributed model? Etc., etc. I'm just throwing out ideas because maybe one of these ideas can turned into a killer app, just like Asterisk itself. Just think of the advantage you would have if you wanted to sell Asterisk against one of the big boys. How much would a fully redundant, HA Asterisk system cost compared to the same thing by Cisco, Avaya, Nortel, NEC... You get the idea. The moral of this post: a little good-natured bashing is just fine, but let's not lose sight of the ultimate goal, which is to keep making Asterisk a better product. Michael, I haven't seen much of a response to this post of yours. That's unfortunate. I was hoping it would spur some constructive conversation. Had you had any response off-list? Doug. ___ --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] FXO for PSTN
At 09:57 PM 6/27/2006, you wrote: 2 TDM400P's with 4 FXO modules each = 8 FXO's = 8 PSTN lines. It's like John said. Very simple maths one would of thought, unless I'm completely off the mark. In which case I do apologise. Probably better off with the TDM2400, 2 fxo boards and the echo can. Ira ___ --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] Work required - modify Asterisk + SEMS
Mike Puchol wrote: Hi all, I am looking for a developer or developers that can implement the following: - Modify an Asterisk server in order to support one inbound RTP and several outbound RTPs, I was thinking SEMS may provide a very good starting point. The idea is to make a PA system over IP. We do *not* want full-duplex audio. - Implement a client in Qt/C++, that allows to send audio to this platform, and plays back audio received from it (Windows-based). We are thinking about Speex for the codec, as there are no royalty issues. Interested parties please reply with your comments, capabilities, so we can start discussing the project. why not setup a listen only meetme for the 'listeners' and talk only for the 'talker'? Jeremy McNamara P.S. Cross posting is not a friendly way to generate discussion, just flames ___ --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] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.
Von L. wrote: plugged in. They work immediately after being plugged in, but they lose the ability shortly thereafter. They can always make outbound calls, but only to real phone numbers, not extensions. They each have NAT routers, and I have triple checked that they have opened/forwarded the correct ports, basically 5060-3 UDP. Once they See the NAT Issues section at http://www.voip-info.org/wiki/view/IAX. (The page is for IAX2, but the NAT issues are relevant for UDP ISP ports too). Basically, some NAT routers forget UDP mappings after a VERY short time (like 30 seconds). Took me a while to figure that out. - Mike ___ --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] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.
Von L. wrote: plugged in. They work immediately after being plugged in, but they lose the ability shortly thereafter. They can always make outbound calls, but only to real phone numbers, not extensions. They each have NAT routers, and I have triple checked that they have opened/forwarded the correct ports, basically 5060-3 UDP. Once they See the NAT Issues section at http://www.voip-info.org/wiki/view/IAX. (The page is for IAX2, but the NAT issues are relevant for UDP SIP ports too). Basically, some NAT routers forget UDP mappings after a VERY short time (like 30 seconds). Took me a while to figure that out. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems compiling Zaptel
asterisk:~# apt-get install kernel-headers-`uname -r` Reading Package Lists... Done Building Dependency Tree... Done kernel-headers-2.4.27-2-386 is already the newest version. Tzafrir Cohen wrote: On Wed, Jun 28, 2006 at 05:22:17PM +0800, Mark Davies wrote: Hi guys, I'm getting the following error when trying to compile zaptel on a debian machine running 2.4.27-3-386. What kernel-headers/kernel-source? I hope you didn't extract a source tarball of 2.4.27/debian and linked it to /usr/src/linux . apt-get install kernel-headers-`uname -r` gcc -g -c -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o zttool.o zttool.c gcc -lnewt zttool.o -o zttool gcc -shared -Wl,-soname,libtonezone.so.1.0 -lm -o libtonezone.so zonedata.lo tonezone.lo make[1]: Leaving directory `/usr/src/zaptel' mkdir -p /dev/zap rm -f /dev/zap/ctl rm -f /dev/zap/channel rm -f /dev/zap/pseudo rm -f /dev/zap/timer rm -f /dev/zap/253 rm -f /dev/zap/252 rm -f /dev/zap/251 rm -f /dev/zap/250 mknod /dev/zap/ctl c 196 0 mknod /dev/zap/transcode c 196 250 mknod: `/dev/zap/transcode': File exists make: *** [devices] Error 1 And when I try to modprobe the X100P card, I get the following. asterisk:/usr/src/zaptel# modprobe wcfxo /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol remove_wait_queue_R5dbd8645 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol __pollwait_R43c77cc3 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol create_proc_entry_Ra52db232 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol proc_mkdir_Rba727c62 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol add_wait_queue_Rf89d8ae0 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol remove_proc_entry_Rf2afedc2 /lib/modules/2.4.27-3-386/misc/zaptel.o: insmod /lib/modules/2.4.27-3-386/misc/zaptel.o failed /lib/modules/2.4.27-3-386/misc/zaptel.o: insmod wcfxo failed This reminds me of the error messages that required the change 2.4.27-1 = 2,4,27-2 . ___ --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 - my cell phone's voicemail sound problems
When I fail to pick up a call from Asterisk to the PSTN to my cell phone and let it go to voicemail, the sound quality is always really bad. When I call my cell phone's voicemail a few minutes later, it's really garbledy and sounds clipped or something. I've tried using Monitor to record the sounds that are being played to my cell's voicemail, and the monitored sound sounds fine when I open it up on my Mac using Quicktime and listen to it. It also sounds fine if I answer the call and listen to it live. Any idea what could be the problem? I'm using BackgroundDetect to figure out when the voicemail prompt finishes, but other than that nothing fancy is going on. thanks, Cory ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems compiling Zaptel
Try shutting off asterisk/zaptel and unloading any zaptel modules (rmmod zaptel, wcfxo, etc) before doing the make install, so udev removes any /dev/ entries associated with them (ie /dev/zap/transcode). If not using udev/devfs, then perhaps unload all zaptel modules, rm -fr /dev/zap, then make install. -tcl. On Wed, 28 Jun 2006, Mark Davies wrote: Hi guys, I'm getting the following error when trying to compile zaptel on a debian machine running 2.4.27-3-386. gcc -g -c -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o zttool.o zttool.c gcc -lnewt zttool.o -o zttool gcc -shared -Wl,-soname,libtonezone.so.1.0 -lm -o libtonezone.so zonedata.lo tonezone.lo make[1]: Leaving directory `/usr/src/zaptel' mkdir -p /dev/zap rm -f /dev/zap/ctl rm -f /dev/zap/channel rm -f /dev/zap/pseudo rm -f /dev/zap/timer rm -f /dev/zap/253 rm -f /dev/zap/252 rm -f /dev/zap/251 rm -f /dev/zap/250 mknod /dev/zap/ctl c 196 0 mknod /dev/zap/transcode c 196 250 mknod: `/dev/zap/transcode': File exists make: *** [devices] Error 1 And when I try to modprobe the X100P card, I get the following. asterisk:/usr/src/zaptel# modprobe wcfxo /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol remove_wait_queue_R5dbd8645 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol __pollwait_R43c77cc3 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol create_proc_entry_Ra52db232 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol proc_mkdir_Rba727c62 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol add_wait_queue_Rf89d8ae0 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol remove_proc_entry_Rf2afedc2 /lib/modules/2.4.27-3-386/misc/zaptel.o: insmod /lib/modules/2.4.27-3-386/misc/zaptel.o failed /lib/modules/2.4.27-3-386/misc/zaptel.o: insmod wcfxo failed Any help is much appreciated. Regards, Mark. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [asterisk-biz] India Routes
We've got a white route w/ VSNL. $0.09 / min, billing is 1/1 prepaid only. SIP or H323 w/ G729 Codec. E-mail me off-list for testing. Thanks, Jon - Original Message - From: Jerry Romney [EMAIL PROTECTED] To: [EMAIL PROTECTED]; asterisk-biz@lists.digium.com; asterisk-dev@lists.digium.com; asterisk-users@lists.digium.com Sent: Wednesday, June 28, 2006 1:44 PM Subject: [asterisk-biz] India Routes Anybody got India Routes at under 8.8 cents? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --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] Standard Sound Files Distortion
I've been noticing lately what seems to be some distortian in the standard asterisk sound files, used for voicemail. These files are stored on the local Asterisk system. When Asterisk plays them, I can hear some cracles and pops. I'd never noticed these until recently. I did a little test. This sounds fine... exten = 1000,1,Answer exten = 1000,n,Wait,1 exten = 1000,n,Playback(digits/p-m) exten = 1000,n,Hangup while this causes some cracks and pops to be heard during the playing of the 'p-m' file. exten = 1000,1,Answer exten = 1000,n,Wait,1 exten = 1000,n,Playback(digits/7) exten = 1000,n,Playback(digits/p-m) exten = 1000,n,Hangup If I replace digits/7 with digits/8, I get the same result. But, as I said, if I just play digits/p-m, I don't hear the distortion. I haven't got a clue what could be causing this. Does anyone else? Doug ___ --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] problem with Asterisk DTMF
Hi,
I used a FXO-Gateway to connect my VoIP to PSTN, using inband mode for
DTMF. It can work properly if I use this dialplan:
exten= 100.,1,Dial(SIP/xxx.xxx.xxx.xxx/111)
111 is a line number in the gateway.
but, I can't get the PSTN number that the caller dialed from the gateway.
So I tried to send the DTMF from asterisk when dialing using this
dialplan:
exten= _1.,1,SIPdtmfMode(inband)
exten= _1.,2,Dial(SIP/xxx.xxx.xxx.xxx/111,,D(${EXTEN:1}))
for some reason, the dialplan above can't work. My assumption is that the
DTMF received by Gateway is different from the one that asterisk sent. I'm
using g711-alaw codec as stated in my sip.conf:
[general]
disallow=all
allow=alaw
...
I already tried using the relaxdtmf but it didn't seem to work.
So now I tried to capture the DTMF that the caller made while dialing but
I can't find a way to do it.
I'm using asterisk 1.2.9.1.
Any solution or a pointer to the problem is welcomed.
Thanks in advance,
Armand
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Re: [Asterisk-Users] Problems compiling Zaptel
On Thu, Jun 29, 2006 at 01:32:36AM +0800, Mark Davies wrote: asterisk:~# apt-get install kernel-headers-`uname -r` Reading Package Lists... Done Building Dependency Tree... Done kernel-headers-2.4.27-2-386 is already the newest version. 2.4.27-*2*-386? Tzafrir Cohen wrote: On Wed, Jun 28, 2006 at 05:22:17PM +0800, Mark Davies wrote: Hi guys, I'm getting the following error when trying to compile zaptel on a debian machine running 2.4.27-3-386. If so: where is this 2.4.27-*3*-386 come from? What kernel-headers/kernel-source? I hope you didn't extract a source tarball of 2.4.27/debian and linked it to /usr/src/linux . apt-get install kernel-headers-`uname -r` gcc -g -c -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -I. -Iinclude -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -o zttool.o zttool.c gcc -lnewt zttool.o -o zttool gcc -shared -Wl,-soname,libtonezone.so.1.0 -lm -o libtonezone.so zonedata.lo tonezone.lo make[1]: Leaving directory `/usr/src/zaptel' mkdir -p /dev/zap rm -f /dev/zap/ctl rm -f /dev/zap/channel rm -f /dev/zap/pseudo rm -f /dev/zap/timer rm -f /dev/zap/253 rm -f /dev/zap/252 rm -f /dev/zap/251 rm -f /dev/zap/250 mknod /dev/zap/ctl c 196 0 mknod /dev/zap/transcode c 196 250 mknod: `/dev/zap/transcode': File exists make: *** [devices] Error 1 And when I try to modprobe the X100P card, I get the following. asterisk:/usr/src/zaptel# modprobe wcfxo /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol remove_wait_queue_R5dbd8645 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol __pollwait_R43c77cc3 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol create_proc_entry_Ra52db232 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol proc_mkdir_Rba727c62 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol add_wait_queue_Rf89d8ae0 /lib/modules/2.4.27-3-386/misc/zaptel.o: /lib/modules/2.4.27-3-386/misc/zaptel.o: unresolved symbol remove_proc_entry_Rf2afedc2 /lib/modules/2.4.27-3-386/misc/zaptel.o: insmod /lib/modules/2.4.27-3-386/misc/zaptel.o failed /lib/modules/2.4.27-3-386/misc/zaptel.o: insmod wcfxo failed This reminds me of the error messages that required the change 2.4.27-1 = 2,4,27-2 . ___ --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 -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Ztdummy and Debian on Intel Macmini
On Wed, Jun 28, 2006 at 06:52:11PM +0200, Olivier wrote: Hi, Is there a way to by-pass the absence of ztdummy on a Debian powered Intel Macmini platform ? The absense of USB? Use kernel 2.6? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Standard Sound Files Distortion
Douglas Garstang wrote: I've been noticing lately what seems to be some distortian in the standard asterisk sound files, used for voicemail. These files are stored on the local Asterisk system. When Asterisk plays them, I can hear some cracles and pops. I'd never noticed these until recently. What I've learned from reading the list, is it usually is a sign of shared IRQs. Just a thought. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Standard Sound Files Distortion
-Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 28, 2006 12:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Standard Sound Files Distortion Douglas Garstang wrote: I've been noticing lately what seems to be some distortian in the standard asterisk sound files, used for voicemail. These files are stored on the local Asterisk system. When Asterisk plays them, I can hear some cracles and pops. I'd never noticed these until recently. What I've learned from reading the list, is it usually is a sign of shared IRQs. Just a thought. Thanks for the reply. I just worked out what it was. I had ulaw copies of all the sound files in the digits/ directory. For some reason, the ulaw files either had the cracks and pops in the recordings, or when asterisk played the ulaw files, it generated the cracks and pops. Doug. ___ --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
