Re: [Asterisk-Users] What do I need to build up DID services?
Greg Hill wrote: On Sat, 25 Dec 2004, Ronald Wiplinger wrote: To complete my project, I would like to setup DIDs in several areas. What do I need to do that? Another Asterisk box or can I use gateways instead? Which hardware can I use? Who has experience? You either set up your own points of presence, or buy service from somebody who already has them set up. So you could install an * with a zap card, a stand-alone spa-3000, or some other device to establish the POP where you want it (and you'll need broadband there too), or you could just buy the service from a VOIP carrier. The latter, IMHO, is better because you won't have to deploy hardware all over. Deployed hardware requires a place to live, somebody to feed it and provide TLC whenever necessary. Usually it's better to keep hardware in as few places as possible to simplify that task.. I am looking for a small device with four FXO and one WAN connection. Simple, so that the cleaning woman can make a hardware reset if necessary. This device should be connnected to my Asterisk box. The box will be used in areas where DIDs are not available yet, and where you not even can make ads for it ;-( It should be cheap, and it should connect either with SIP or IAX. No FXS is needed !!! If you don't like the VOIP delivery, telcos for years have offered market expansion lines which give you a number in a remote rate center which automatically forwards to your real number. (funny thing is, when I talked to Qwest about that a month ago, this forwarding service cost more than regular residential phone service!) Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linux Distribution
Seth Ueland Chancy wrote: What are the known distributions of Linux with which Asterisk is known not to work? To answer the real question which is on the back of your head, unless you're lucky you'll probably have to do a lot of fiddling around no matter which distro you choose to get * to work... Probably your best bet is debian + 2.4 kernel + X100P card + apt-get install asterisk Cheers, Jean-Michel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialogic Support
I am a newbie to asterisk pbx. I got a dialogic card with 2 ports. Can any one tell whether asterisk supports dialogic cards? Update me if you have info about the drivers installation and support. Thanks Regards V.Venu ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Tie web application to VOIP
Hi, -Original Message- I want to tie my web application (built using .NET + MS SQL Server) into a VOIP service so that users can call each other. I want them to interface with my application's username system. On the receiving user's end, he can either receive the call using a VOIP phone, or windows application (like skype). I would use Skype's API, but it appears I need to use their username system, and not my own. My question is, what software/hardware solutions would I need to do this? Any suggestions/feedback would be greatly appreciated. Btw, I was told that Asterisk + SER would do the trick. However, I'm a newbie to the world of VOIP. If someone can give me some tips/hints, it would be great. Okay, lets cut it out with all the database and OS flamewars. In general, and almost entirely independent of what database you are using, you could do the following: Have a linux machine with your VoIP setup (Asterisk would be my choice of the moment) and create a cronjob on there that will check if the database has modified entries (i.e. through the ODBC layer). If so, you create (partial) configuration files that suit your needs and reload the asterisk machine. Only issue is: You will need to have cleartext passwords in your database. I am aware this is a bad design issue, but the only workaround I can come up with is not acceptable to me: If you can enforce md5 authentication on all your users, you'd have enough with the md5 hash in the database. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transcript of sound files?
I want to record new sound files in different languages, but I need the text files of the English ones, which I would use as basic. Since some languages already exists, I believe such a list should be exist, but where? I am planning to make Chinese and Tagalog sound files. If therese are existing, I can save the work too. bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Lucent APX8100 Universal Gateway
we're evaluating the use of a Lucent APX8100 E3/SS7 to SIP gateway for use in conjunction with asterisk, serving something like 4000+ lines. does anyone have experience with the APX8100 and it's integration with SIP on asterisk ? does the APX8100 handle SS7-SIP signalling well enough to be used ? any anecdotes would be well appreciated. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do I need to build up DID services?
On 25/12/2004 16:27 Ronald Wiplinger said the following: I am looking for a small device with four FXO and one WAN connection. Simple, so that the cleaning woman can make a hardware reset if there're a number of FXO/SIP gateways you can consider. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transcript of sound files?
On 25/12/2004 16:48 Ronald Wiplinger said the following: I want to record new sound files in different languages, but I need the text files of the English ones, which I would use as basic. Since some languages already exists, I believe such a list should be exist, but where? see http://www.voip-info.org/wiki-Asterisk+sound+files and http://www.voip-info.org/wiki-Asterisk+sound+files+additional -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to use firefly with Asterisk?
I have installed Firefly, but I cannot figure out how to use it with Asterisk. I have seen the settings in Asterisk, but I do not see any settings in Firefly. I need a light bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dynamic extensions without using DynExtenDB?
Hi,
Am using Asterisk to call
peoples phone as part of a service of my website. It will call people for
various things...one of them to tell people sports scores. I am using
several sound files to piece together a dynamic message saying who played and
what the score was. The problem is that I can hardcode the sound files
that are neededto play and it works fine, but I cannot hardcode the
extensions.conf file with the new sound files to play for every game. I
need a way to create dynamic extensionssowhen it dials out, it will
beplaying back the correct sound files for the current game
beingplayed.
I've looked at the DynExtenDB,
but that wont work well because every time a call is initiated, it loads the
extension. Thats not practical because ifthe system needs to call
500 people right at the end of the game, its going to take too much overhead
just to load and unload the extension for every call.
I am thinking now just to load
the extensions through MySQL andhave an "extensions reload" command
sentthe CLI for Asterisk so my backend system can update mysql and then
reload the extensions on Asterisk when a call needs to be made with new sport
scores.
I think that idea will
work...but does anybody know of a better way to do it?
P.S. I know I can reload the extensions by
typing "extensions reload" in the CLI, but is there a command line command I can
run that will do it without having to login? It will make my scripting job
a lot easier if I can just have a perl scrip run something like
system("/usr/sbin/asterisk -r extensions reload")
Thanks!
Gabe
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[Asterisk-Users] Can Asterisk handle calls that get picked up by answering machines?
Just wondering because right now I can have it call my phone and play a message, but if I dont answer it eventually goes to voice mail. It always leaves a voicemail and when I listen to it its always the last few seconds of my message that I had Asterisk play. How do I get Asterisk to pause and wait to playback *IF* its an answering machine or voicemail? Gabe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P No Interrupts
Hi all, Just got a brand new server and a Digium TE410P. I get the sequential (knight rider) lights before loading the zaptel driver. As soon as I load the driver all loghts go off. It appears the card is not generating interrupts. [EMAIL PROTECTED] ~]# cat /proc/interrupts CPU0 0: 111005341IO-APIC-edge timer 1: 9IO-APIC-edge i8042 8: 1IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 12: 66IO-APIC-edge i8042 14: 7870IO-APIC-edge ide0 185: 0 IO-APIC-level t4xxp 193: 26141 IO-APIC-level cciss0 201:1139611 IO-APIC-level eth0 NMI: 0 LOC: 111010062 ERR: 0 MIS: 0 [EMAIL PROTECTED] ~]# Have also tried replacing the card, changinf PCI slots and messing with the BIOS all with the same result. If anyone can help I would be very grateful.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bandwidth, computer power
I want to use MY asterisk server to help some people to get MULTIPLE gateways to their VoIP phone. E.g.UK, US, Canada DIDs, registered to my asterisk server, with dedicated dialing to my friend. I believe that the RTP (voice streams) are going directly from the DIDs to my friends, and will not use up my bandwidth. I believe that however, if he use my mailbox it will use my bandwidt, as well he will use twice the bandwidth, if he use a conference call with a third party. Besides the confirmation about the above, I would like how much bandwidth I will need to handle such calls for my friend? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Automatic calls
I am looking for an automatic call out system. - NOT FOR SPIT !!! I remember my teacher who said to collect money, use your wife!!! She usually can pretend to know nothing about the business, but her job is to get the invoices paid. He was very right with that! I used that idea in the past to send automatical faxes about delayed payments of my customers, and it worked fine. You always could explain your best customers, that this is an automatic program and does not know our special relationship, but the invoice is overdue though. Usually I got than paid. Now I want to use the same idea with the phone. I want to call the customer, and tell him that the bill is overdue! Which invioice number, date, amount!!! All taken from a database. Is such an application available? I get sometimes phone calls from insurances to my birthday, I could use such a program to modify it to that. bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dynamic extensions without using DynExtenDB?
On Sat, 25 Dec 2004, Gabriel Afana wrote: I've looked at the DynExtenDB, but that wont work well because every time a call is initiated, it loads the extension. Thats not practical because if the system needs to call 500 people right at the end of the game, its going to take too much overhead just to load and unload the extension for every call. I am thinking now just to load the extensions through MySQL and have an extensions reload command sent the CLI for Asterisk so my backend system can update mysql and then reload the extensions on Asterisk when a call needs to be made with new sport scores. Have you considered using agi instead of a lot of extensions? Especially with fastagi it may be an easier route to go. P.S. I know I can reload the extensions by typing extensions reload in the CLI, but is there a command line command I can run that will do it without having to login? It will make my scripting job a lot easier if I can just have a perl scrip run something like system(/usr/sbin/asterisk -r extensions reload) asterisk -rx extensions reload Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ALERT_INFO issue CVS-HEAD-12/24/04
On Sat, 25 Dec 2004, John Bittner wrote: Anyone having any problems with CVS-HEAD-12/24/04-15:59:15 and ALERT_INFO I have a system setup with polycom phones configured to auto answer on internal calls. When we upgraded to the latest CVS the auto answer stopped working. My dialplan has not changed. I did a sip debug and I dont see the alert-info tag in any of the sip traces. This is a what I have in my dialplan. exten = 207,1,SetVar(ALERT_INFO=Ring Answer) exten = 207,2,Dial(SIP/207) exten = 207,3,Hangup This has been covered onm asterisk-users already. The syntax for passign ALERT_INFO has changed. Set the variable _ALERT_INFO instead of ALERT_INFO. The new, outgoing, channel will inherit ALERT_INFO then. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] About CallBack function
Hello , Do anyone know how Asterisk support CallBack function ? Are there some documents about CallBack of Asterisk? Thanks! Comer [EMAIL PROTECTED] 2004-12-25 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] About CallBack function
Hello , Do anyone know how Asterisk support CallBack function ? Are there some documents about CallBack of Asterisk? Thanks! Comer [EMAIL PROTECTED] 2004-12-25 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] About CallBack function
Hello , Do anyone know how Asterisk support CallBack function ? Are there some documents about CallBack of Asterisk? Thanks! Comer [EMAIL PROTECTED] 2004-12-25 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] About CallBack function
Hello , Do anyone know how Asterisk support CallBack function ? Are there some documents about CallBack of Asterisk? Thanks! Comer [EMAIL PROTECTED] 2004-12-25 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to connect two Asterisks as secure as possible without too much additional bandwidth ?
Hi, I plan to connect to remote Asterisk that will terminate calls to ISDN primary channel. I'd certainly like to secure this type of service, so would kindly ask for any advice on how to secure this authentication as much as reasonably possible. Since there is long IP route I guess VPN will take too much additional bandwidth... Regards, Robert. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can Asterisk handle calls that get picked up by answering machines?
Gabriel Afana wrote: Just wondering because right now I can have it call my phone and play a message, but if I dont answer it eventually goes to voice mail. It always leaves a voicemail and when I listen to it its always the last few seconds of my message that I had Asterisk play. How do I get Asterisk to pause and wait to playback *IF* its an answering machine or voicemail? Gabe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You can hack the record application to wait for silence. -- Todd Lieberman mailto:[EMAIL PROTECTED] http://tlsolutions.net 215.495.0030 (p) 215.495.0031 (f) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Help on Register message with Proxy-Authorization
i have gone through RFC 2617 and able to run the test program but problem i that when i try these values this is not matching with my asterisk proxy response can any one tell me what should be pszMethod here i dont have Qop here char * pszNonce = 539b02d7; char * pszCNonce = ; char * pszUser = 3000; char * pszRealm = asterisk; char * pszPass = kamran; char * pszAlg = md5; char szNonceCount[9] = ; char * pszMethod = ; char * pszQop = ; char * pszURI = sip:192.168.0.11; HASHHEX HA1; HASHHEX HA2 = ; HASHHEX Response; DigestCalcHA1(pszAlg, pszUser, pszRealm, pszPass, pszNonce, pszCNonce, HA1); DigestCalcResponse(HA1, pszNonce, szNonceCount, pszCNonce, pszQop, pszMethod, pszURI, HA2, Response); printf(Response = %s\n, Response); __ Do you Yahoo!? Yahoo! Mail - Easier than ever with enhanced search. Learn more. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do I need to build up DID services?
On Sat, Dec 25, 2004 at 04:27:52PM +0800, Ronald Wiplinger wrote: Greg Hill wrote: I am looking for a small device with four FXO and one WAN connection. Simple, so that the cleaning woman can make a hardware reset if necessary. This device should be connnected to my Asterisk box. The box will be used in areas where DIDs are not available yet, and where you not even can make ads for it ;-( It should be cheap, and it should connect either with SIP or IAX. No FXS is needed !!! I'm not sure, but I think your best price/performance solution might be a very small linux system (1U or smaller) with one Digium TDM400P+4*FXO daughtercards. At least the the size point of 4*FXO, this may turn out to be your winner. If it boots from flash, your cleaning lady can always apply power cycle for anything you can't fix via ssh... By using this solution, you would also get the ongoing bandwidth benefit of being able to trunk all 4 FXO calls out in a single IAX frame. This will save you money on an on-going basis if your bandwidth costs are significant. Perhaps Soekris would be a good box for this solution, but there may be even smaller devices with the needed 1*PCI interface, I don't know. Regards, -Dorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linux Distribution
On Sat, Dec 25, 2004 at 12:29:21PM +, Jean-Michel Hiver wrote: Seth Ueland Chancy wrote: Probably your best bet is debian + 2.4 kernel + X100P card + apt-get install asterisk Cheers, Jean-Michel. I can also confirm that * works fine on Debian w/2.6.10-rc2-mm3 for the adventurous :) -Dorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialogic Support
Venu V wrote: I am a newbie to asterisk pbx. I got a dialogic card with 2 ports. Can any one tell whether asterisk supports dialogic cards? Update me if you have info about the drivers installation and support. This has been discussed over and over and over again on this mailing list. See http://www.asteriskpbx.org/index.php?menu=hardware for a list of Dialogic cards that are supported. The drivers are not free. Contact Digium for pricing information. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transcript of sound files?
Ronald Wiplinger wrote: I want to record new sound files in different languages, but I need the text files of the English ones, which I would use as basic. Since some languages already exists, I believe such a list should be exist, but where? See sounds.txt in the Asterisk source code directory. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fw: [digium.com #12961] T100P as bandwidth
Quoting TC [EMAIL PROTECTED]: I agree -- I must admit I added the IAXy to pad out my argument -- my main beef is with the TDM4XXP card/driver. Nonsense the IAXy not only has some driver / hardware issues but the feature set make it unuseable in profession corporate enviroments no echo can no cpu for std codec like g723/g729/ilbc, only pcm/ulaw No DNS, bootp only bad provisioning IAXy has no security Not configurable via http single port price point $99, (sipura 2 port 75us), or 1 port $64 Oh wow. I didn't know the IAXy lacked all those features. I can get a Linksys PAP2-NA for $50 that has g729, dhcp, http config, echo can, etc..etc..I wanted to use the IAXy in deployment to work around NAT issues using IAX but if the hardware doesn't support some of the most basic features... Matthew This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linux Distribution
On December 25, 2004 07:29 am, Jean-Michel Hiver wrote: To answer the real question which is on the back of your head, unless you're lucky you'll probably have to do a lot of fiddling around no matter which distro you choose to get * to work... I have no idea what fiddling you're talking about -- Asterisk will run just fine on any distro I can think of. It certainly takes no fiddling on Slackware. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to use firefly with Asterisk?
You should use Firefly Third party edition I'm using it in IAX2 and it works fine bye, M. - Original Message - From: Ronald Wiplinger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 25, 2004 6:29 AM Subject: [Asterisk-Users] How to use firefly with Asterisk? I have installed Firefly, but I cannot figure out how to use it with Asterisk. I have seen the settings in Asterisk, but I do not see any settings in Firefly. I need a light bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New TDM11B. FXS detach! We failed: 5
I just got the new developer TDM11B, but I got some problems with it. Since Digium is on vacation, I figured I'd ask here first: I installed the TDM11B, but have not attached any phone lines, yet. I just want to work with the demo over SIP first. But here's the story, after installing the card on a Dell SC420 (further identifying info comes from dmesg): FreeBSD 5.3-RELEASE #0: Fri Nov 5 04:19:18 UTC 2004 [EMAIL PROTECTED]:/usr/obj/usr/src/sys/GENERIC Timecounter i8254 frequency 1193182 Hz quality 0 CPU: Intel(R) Celeron(R) CPU 2.53GHz (2527.01-MHz 686-class CPU) when init runs /usr/local/etc/rc.d/zaptel.sh I get the following info: Zapata Telephony Interface Registered on major 196 FXS device: vendor=e159 device=1 subvendor=b100 wcfxs0: Wildcard TDM400P REV E/F port 0xdc00-0xdcff mem 0xdfaff000-0xdfaf irq 18 at device 2.0 on pci4 FXS Attach for wcfxs0: deviceID : 0xe159 wcfxs0: [GIANT-LOCKED] Freshmaker version: 71 Freshmaker passed register test Module 0: Not installed Module 1: Not installed Module 2: Not installed Module 3: Not installed Freed a Wildcard FXS detach! We failed: 5 device_attach: wcfxs0 attach returned 5 I don't get why it says we failed. I thought maybe it was the order that the modules are loaded (currently they are: zaptel, wcfxo, wcfxs) but I get the same error if I reverse wcfxo and wcfxs. I get the following devices created in /dev/zap, but none of them are configured (i.e. if I `cat /dev/zap/timer` I get device not configured). crw-rw-r-- 1 root wheel 230, 254 Dec 25 12:46 channel crw-rw-r-- 1 root wheel 230, 0 Dec 25 12:46 ctl crw-rw-r-- 1 root wheel 230, 255 Dec 25 12:46 pseudo crw-rw-r-- 1 root wheel 230, 253 Dec 25 12:46 timer I can make asterisk run, and I can connect to it using a software SIP phone. I can even hear the demo, but it is wa choppy. So I figure that the choppiness will diminish once I can get the FXS module to load. So ... how do I get wcfxs to load? Thanks! lane ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New TDM11B. FXS detach! We failed: 5
I just got the new developer TDM11B, but I got some problems with it. Since Digium is on vacation, I figured I'd ask here first: I installed the TDM11B, but have not attached any phone lines, yet. I just want to work with the demo over SIP first. But here's the story, after installing the card on a Dell SC420 (further identifying info comes from dmesg): FreeBSD 5.3-RELEASE #0: Fri Nov 5 04:19:18 UTC 2004 [EMAIL PROTECTED]:/usr/obj/usr/src/sys/GENERIC Timecounter i8254 frequency 1193182 Hz quality 0 CPU: Intel(R) Celeron(R) CPU 2.53GHz (2527.01-MHz 686-class CPU) when init runs /usr/local/etc/rc.d/zaptel.sh I get the following info: Zapata Telephony Interface Registered on major 196 FXS device: vendor=e159 device=1 subvendor=b100 wcfxs0: Wildcard TDM400P REV E/F port 0xdc00-0xdcff mem 0xdfaff000-0xdfaf irq 18 at device 2.0 on pci4 FXS Attach for wcfxs0: deviceID : 0xe159 wcfxs0: [GIANT-LOCKED] Freshmaker version: 71 Freshmaker passed register test Module 0: Not installed Module 1: Not installed Module 2: Not installed Module 3: Not installed Freed a Wildcard FXS detach! We failed: 5 device_attach: wcfxs0 attach returned 5 I don't get why it says we failed. I thought maybe it was the order that the modules are loaded (currently they are: zaptel, wcfxo, wcfxs) but I get the same error if I reverse wcfxo and wcfxs. I get the following devices created in /dev/zap, but none of them are configured (i.e. if I `cat /dev/zap/timer` I get device not configured). crw-rw-r-- 1 root wheel 230, 254 Dec 25 12:46 channel crw-rw-r-- 1 root wheel 230, 0 Dec 25 12:46 ctl crw-rw-r-- 1 root wheel 230, 255 Dec 25 12:46 pseudo crw-rw-r-- 1 root wheel 230, 253 Dec 25 12:46 timer I can make asterisk run, and I can connect to it using a software SIP phone. I can even hear the demo, but it is wa choppy. So I figure that the choppiness will diminish once I can get the FXS module to load. So ... how do I get wcfxs to load? I have a tdm04b, but not with any fxs modules in it. So, I'll guess that your asterisk code is rather old since the tdm card has been using wctdm for some time. The driver wcfxs does not exist in the current cvs head code. It would be helpful to know which linux kernel (or distro) you're using, and which asterisk version you've got installed. Without any of that info, you might try: modprobe wctdm ztcfg Do you have /etc/zaptel.conf with fxsks=1 fxoks=4 (or something like that)? There has been a fair amount of discussion relative to configuring that card on the list. Might be helpful to google for some of it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID - TE405P - Telstra Onramp 10 - Australia
David, This ended up being the problem, it was enabled and is working as expected now. Thank you, Nathan. I know this might be a basic answer, but have you confirmed that CID is enabled and working on the onramp? I know when I dealt with T for an OnRamp 30 18months ago it was ordered with CID enabled but did not work for weeks when it should have. When T was chalanged about the problem it was found out that it was not enabled :( They enabled it and all the problems went away. Might be worth a thought anyway. David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New TDM11B. FXS detach! We failed: 5
On December 25, 2004 02:07 pm, Lane wrote: I can make asterisk run, and I can connect to it using a software SIP phone. I can even hear the demo, but it is wa choppy. So I figure that the choppiness will diminish once I can get the FXS module to load. Remove the card entirely and run the demo -- how is the audio? -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New TDM11B. FXS detach! We failed: 5
Lane wrote: I just got the new developer TDM11B, but I got some problems with it. Since Digium is on vacation, I figured I'd ask here first: I installed the TDM11B, but have not attached any phone lines, yet. I just want to work with the demo over SIP first. But here's the story, after installing the card on a Dell SC420 (further identifying info comes from dmesg): FreeBSD 5.3-RELEASE #0: Fri Nov 5 04:19:18 UTC 2004 [EMAIL PROTECTED]:/usr/obj/usr/src/sys/GENERIC Timecounter i8254 frequency 1193182 Hz quality 0 CPU: Intel(R) Celeron(R) CPU 2.53GHz (2527.01-MHz 686-class CPU) I doubt Digium will provide support for using *BSD with Digium hardware. Your best bet might be the asterisk-bsd mailing list or try using Linux. Not a lot of people use Digium hardware with *BSD so your pool of people that might be able to help you is significantly smaller than the pool of people using Digium hardware on Linux. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Record() problem
That is what I used :) except I got it from the another page in the Wiki I
think.. I just changed the sound file references to a sound file that
existed on my side.
After using this example I got the error when * gets to the record line of
extensions.conf:
WARNING[3293201]: app_record.c:117 record_exec: No extension found
Thanks!
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, December 24, 2004 10:38 PM
Subject: RE: [Asterisk-Users] Record() problem
http://bugs.digium.com/bug_view_page.php?bug_id=0002905
Refer to my example on that bug note.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Me
Sent: Friday, December 24, 2004 11:06 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-
Commercial Discussion
Subject: Re: [Asterisk-Users] Record() problem
It was executed from the dial plan within extensions.conf and I did not
hard
code the IAX2/[EMAIL PROTECTED]/5 in there. I will paste the exact text
below
from my extensions.conf which I really should have done the first time
:)
sorry..
I didn't include the Macro but that's not where it's blowing up. Any
help
would be appreciated.
Happy Holidays to all!
*From extensions.conf*
; 1100 - Test call whisper type thing
;exten = 1100,1,Wait(0.2)
;exten = 1100,2,Playback(say-name)
;exten = 1100,3,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH})
;exten = 1100,4,Record(${SCREEN_FILE}.gsm,6,25)
;exten = 1100,5,Dial(SIP/1100,60,gM(screen^${SCREEN_FILE}))
;exten = 1100,6,Voicemail([EMAIL PROTECTED])
***End
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message -
From: Bill Seddon [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, December 24, 2004 3:00 AM
Subject: RE: [Asterisk-Users] Record() problem
You syntax for the command is incorrect. See
http://www.voip-info.org/wiki-Asterisk+cmd+record.
Record is an application to be executed from within the dialplan. So
the
channel it will record is implicit and cannot be explicitly stated as
one
of
the parameters.
If you want to originate and record a call automatically, you will
have
to
do this via AGI.
Bill Seddon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Me
Sent: December 24, 2004 6:38 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Record() problem
Any idea why this:
Record(IAX2/[EMAIL PROTECTED]/5, /tmp/whatever.gsm|6|25)
Would result in this:
WARNING[3293201]: app_record.c:117 record_exec: No extension found
Thanks!
--
Start Your Own ISP!
http://www.YourOwnISP.com
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RE: [Asterisk-Users] Premature DRQ
Didn't see any responses... I looked at the logs more closely and I see a message that says releaseCompleteReason - destinationRejection. This is showing around 60 seconds when the DRQ occurs and the connection is broken. -Original Message- From: Huddleston, Robert [mailto:[EMAIL PROTECTED] Sent: Thursday, December 23, 2004 2:08 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Premature DRQ I have a problem where an Asterisk server is sending a premature DRQ... Not sure why.. Here's the setup - Asterisk using inAccess networks H323 replacement channel driver Connecting to a Lucent iMerge... The call connects fine - I get the out of the box greeting - but after exactly one Minute - the call terminates. I have had this problem on multiple different Asterisk configs... I'm assuming the trouble is coming from the Lucent iMerge product... I have the log as seen from the Lucent iMerge - and thinking that the line before DRQ Is causing the trouble. 213 2004-12-23 13:46:34 2004-12-23 14:37:17 H.323 Line:1-1401 302224161067.62.108.117RX LogicalChannel:101 Codec: g711Ulaw64k @ 20ms Ports: RTP:0 RTCP:10033 214 2004-12-23 13:47:34 2004-12-23 14:38:16 Activity Line:1-1401 302224161067.62.108.117 Sending Supy On: LogDS0.26.15 Type = Onhook: 215 2004-12-23 13:47:34 2004-12-23 14:38:16 H.323 Line:1-1401 302224161067.62.108.117 rcv DRQ, Call ID: 9101839cf281d54fb693a0d89cb5cfb4 220 2004-12-23 13:47:46 2004-12-23 14:38:30 GR-303Line:1-1401 302224161067.62.108.117 Received GR303 message: DISCONN cause=16.00x, A CRV=1401 typ=45x Len= 10: 4f 02 2b c8 45 08 03 82 221 2004-12-23 13:47:46 2004-12-23 14:38:30 Activity Line:1-1401 302224161067.62.108.117 Sending Supy On: LogDS0.26.15 Type = IdleCode: 222 2004-12-23 13:47:46 2004-12-23 14:38:30 Resource Line:1-1401 302224161067.62.108.117 Released resource PhysDS0.1.10.2.15 223 2004-12-23 13:47:46 2004-12-23 14:38:30 Resource Line:1-1401 302224161067.62.108.117 Released resource DSPchannel.1.8.1.32 224 2004-12-23 13:47:46 2004-12-23 14:38:30 Activity Line:1-1401 302224161067.62.108.117 Deleting Media Path Robert A. Huddleston, KF4BYY Cavalier Telephone LLC. (Desk) 804.422.4401 (Cell) 804.400.3686 [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bri-stuff + TDM 2-Port FXS 2 Port FXO Card
Hi Mary Christmas to y'all ! I am trying to configure * with one TDM and one ISDN and Bri-stuff, all is ok with only TDM but when add ISDN I get following error: I have tried to switch fxs/fxo but then I get error also without ISDN, card is configured from Digium with FXS(green) modules are closest to the bracket. ZT_CHANCONFIG failed on channel 1: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? loadzone=no defaultzone=no fxoks=1-2 # Make sure that the FXS(green) modules are closest to the bracket fxsks=3-4 # This is for the FXO module(s) because it uses FXS span=2,1,3,ccs,ami bchan=5-6 # Old value: 1-2 dchan=7# Old value: 3 Thank you ! HB Norway ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] patch to build h323 without recompiling pwlib, ...
Heya, I changed the Makefile of the h323-channel-makefile (I downloaded cvs of a couple of hours ago) so that I don't have to rebuild pwlib and openh323, but can use the precompiled versions. I'm using pwlib 1.8.3 and openh323 1.15.2. There aren't many changes. I replaced OPENH323DIR with OPENH323INC ,which points to /usr/include/openh323 for me and OPENH323LIB, which points to /usr/lib for me. The variable OPENH323DIR is still there after teh patch, but can be removed I think. That's it I think. I just assigned the variable PWLIBDIR, the value /usr , that worked for me. The Makefile I'm talking about is: channels/h323/Makefile Anyway, this way I don't have to recompile those libraries, but can use the installed ones ... I'll attach the patch and hope it will get through, but the above explanation says it all I think. greetz, Michel Brabants --- Makefile2004-12-25 19:56:45.0 +0100 +++ ../../../../Makefile-h323 2004-12-25 19:55:59.0 +0100 @@ -1,11 +1,13 @@ # include the Makefile of OpenH323 ifndef OPENH323DIR -OPENH323DIR=$(HOME)/openh323 +OPENH323DIR=/ +OPENH323INC=/usr/include/openh323 +OPENH323LIB=/usr/lib endif ifndef PWLIBDIR -PWLIBDIR=$(HOME)/pwlib +PWLIBDIR=/usr endif ifndef ASTERISKDIR @@ -41,7 +43,7 @@ CFLAGS += -D_REENTRANT -D_GNU_SOURCE CFLAGS += -I../../include CFLAGS += -I$(PWLIBDIR)/include -CFLAGS += -I$(OPENH323DIR)/include -Wno-missing-prototypes +CFLAGS += -I$(OPENH323INC) -Wno-missing-prototypes all: depend libchanh323.a @@ -64,13 +66,13 @@ endif chan_h323.so: - $(CXX) -g -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L$(PWLIBDIR)/lib -lpt_linux_x86_r -L$(OPENH323DIR)/lib -lh323_linux_x86_r -L/usr/lib $(CHANH323LIB) + $(CXX) -g -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L$(PWLIBDIR)/lib -lpt_linux_x86_r -L$(OPENH323LIB) -lh323_linux_x86_r -L/usr/lib $(CHANH323LIB) chan_h323_d.so:chan_h323.o ast_h323.o - $(CXX) -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L$(PWLIBDIR)/lib -lpt_linux_x86_d -L$(OPENH323DIR)/lib -lh323_linux_x86_d -L/usr/lib $(CHANH323LIB) + $(CXX) -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L$(PWLIBDIR)/lib -lpt_linux_x86_d -L$(OPENH323LIB) -lh323_linux_x86_d -L/usr/lib $(CHANH323LIB) chan_h323_s.so:chan_h323.o ast_h323.o - $(CXX) -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L$(PWLIBDIR)/lib -lpt_linux_x86_r_s -L$(OPENH323DIR)/lib -lh323_linux_x86_r_s -L/usr/lib $(CHANH323LIB) + $(CXX) -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L$(PWLIBDIR)/lib -lpt_linux_x86_r_s -L$(OPENH323LIB) -lh323_linux_x86_r_s -L/usr/lib $(CHANH323LIB) clean: rm -f *.o *.so core.* libchanh323.a .depend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] safe_asterisk script contains error?
Heya, I was just trying out the safe_asterisk-script. I think that the 2 asterisk-commands in the script always need the -f-option, else the script doesn't really do what it is intended for I think ... except only on startup. greetz, Michel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where to get a Polycom IP500 in the UK?
I've been looking at Asterisk for a while now, and want to set up a small installation at my house. I did originally want to get a Cisco IP phone, but due to price and it not being easy to obtain the SIP firmware I have decided again them. I am very interested in the Polycom IP500. I can find many places in the USA that sell it but nowhere in the UK. Also can anyone recommend some good VoIP suppliers in the UK? Thanks, Martin Mitchell ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Tie web application to VOIP
Steven, Just a quick reminder, MS SQL on Windows is hands down the best performing transact SQL database on the planet, and Oracle on Windows is a close #2. Some might argue that Oracle is #1 and MS is #2. Anyone that argues any Linux SQL db even comes close in performance better provide some evidence to back their argument. I think I would cite Oracle. They say their database runs much better on Linux than on Windows. If Oracle on Windows is arguably better than MS SQL, Oracle on linux must be a hands down winner. Actually, Oracle just posted a claim that they are the world record holder, but they did so using 12 AMD Opterons, interestingly enough the last time they boasted about performance it was on Intel CPUs. Maybe it is AMD that is the real winner... http://www.oracle.com/solutions/performance_scalability/tpch1tb_hplinux_ 1004.html The last time I used MS SQL was about 5 years ago, but it was hands down the biggest heap of crap in the database world back then. I guess it has improved, but if you think its the best out there I guess you haven't tried very much stuff. 5 years ago, must have been SQL 6.5 or 7.0, and yes, it has improved, in fact for a number of years SQL 2000 was 2x faster than Oracle on any platform. The asp.net SQL database providers for MS SQL and Oracle SQL are highly optimized direct socket interfaces to the SQL server. (no odbc crap!) I think you've been drinking the Koolade. Actually, I prefer scotch, but then again, I am the one that defends windows, so what do I know. Maybe SAP, PeopleSoft, Siebel, and most other Enterprise software vendors are drunken idiots too. I make a living implementing these solutions, and I would love to see a Linux alternative, but to date it does not exist. Oracle on Linux is nice, but sure is a bitch integrating the security with Windows desktops. Now that we are completely off topic, lets get back to what this was about in the first place, The person that started this thread was asking about the possibility of using ASP.net/SQL as a front end to *. Very possible, and an added benefit would be you might be able to use a single database for your user interface, accounting, * and SER configs, marketing, billing, etc since MS SQL is supported by more applications than any other database on earth. This is the direction we are moving towards, and it does look promising. Any suggestions for a Linux based accounting and CRM package? Damon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Regex in number dialed
Hello Didn't find any information in the wiki. Regex only refers to the dialing syntax I'd like to do something like: exten = _8001133[1-5,7-9]XX.,1,Dial(SIP/france-gateway,60,tr) is there a possibility? right now I've had to enter all possible choice like: exten = _80011331XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011332XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011333XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011334XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011335XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011337XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011338XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011339XX.,1,Dial(SIP/france-gateway,60,tr) Thank you in advance Cheers Jean-Yves ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Regex in number dialed
exten = _8001133[12345789]XX.,1,Dial(SIP/france-gateway,60,tr) All those lines can be taken to that. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jean-Yves Avenard Sent: Saturday, December 25, 2004 7:43 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Regex in number dialed Hello Didn't find any information in the wiki. Regex only refers to the dialing syntax I'd like to do something like: exten = _8001133[1-5,7-9]XX.,1,Dial(SIP/france-gateway,60,tr) is there a possibility? right now I've had to enter all possible choice like: exten = _80011331XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011332XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011333XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011334XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011335XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011337XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011338XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011339XX.,1,Dial(SIP/france-gateway,60,tr) Thank you in advance Cheers Jean-Yves ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Tie web application to VOIP
Just a quick reminder, MS SQL on Windows is hands down the best performing transact SQL database on the planet, and Oracle on Windows is a close #2. Some might argue that Oracle is #1 and MS is #2. Anyone that Does Oracle have a decent-featured free version of their db software? That was my original point, and where MS SQL 2005 is quite in the lead (limited only to 1GB of RAM, 4GB DB, and 1 CPU). -Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Predictive dialer
Hello: I need to install and use a predictive dialer at my * box Can somebody recomend me any ? Any kind of point of view would be appreciated Thanks in advance --- Ing. Julio Alvarez Tejera Unix Trends *BSD, Solaris Linux --- extremely stable systems ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VM_CALLERID (how to get name+number)
I'd like to get VM_CALLERID to include number in addition to name since often when calls come from cell lines or various other, the name is just a city, state and the number would be more usefull. Is there a way to get the number in the VM_CALLERID string, or is there a second variable I can use in formatting email vmail notifications to get the number? Regards, -Dorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Special Problem in Australia ??
On Fri, 24 Dec 2004 12:17:50 +1000, Gary [EMAIL PROTECTED] wrote: Hi folks, this is specifically directed to Australia Asterisk users.. We are having a roblem with x100p 's when dialing STD. Upon receipt of the approximately the 5th (out of the ten) PIP's asterisk will hang up Now I am wondering if others are suffering the same problem ?? Any ideas ?? (it might exist on other cards, but so far I have only noticed the problem on x100p's). Gary I use a Telstra BRI now, but when I was using the X100P cards I had the problem. It was fixed by increasing the busycount value in zapata.conf. Anyway, I see in a later message that you've fixed the problem now. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Special Problem in Australia ??
Just out of interest, what BRI card are you with Asterisk? On Sun, 26 Dec 2004 16:26:37 +1100, Shaun Ewing [EMAIL PROTECTED] wrote: On Fri, 24 Dec 2004 12:17:50 +1000, Gary [EMAIL PROTECTED] wrote: Hi folks, this is specifically directed to Australia Asterisk users.. We are having a roblem with x100p 's when dialing STD. Upon receipt of the approximately the 5th (out of the ten) PIP's asterisk will hang up Now I am wondering if others are suffering the same problem ?? Any ideas ?? (it might exist on other cards, but so far I have only noticed the problem on x100p's). Gary I use a Telstra BRI now, but when I was using the X100P cards I had the problem. It was fixed by increasing the busycount value in zapata.conf. Anyway, I see in a later message that you've fixed the problem now. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk + Voice Modem
Dear All, Can somebody point me to resources about how to configure asterisk with a voice modem? I am having a soft-modem modem which is Intel AMR and using slmodemd soft-driver. Thanks -- There are 10 kinds of people in the world: Those who understand binary and those who don't... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New TDM11B. FXS detach! We failed: 5
On 26/12/2004 03:35 Eric Wieling aka ManxPower said the following: I doubt Digium will provide support for using *BSD with Digium hardware. Your best bet might be the asterisk-bsd mailing list or try using i feel that digium should consider supporting BSD formally, given that the number of people using digium/asterisk on *BSD is increasing. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Regex in number dialed
Brian West wrote: exten = _8001133[12345789]XX.,1,Dial(SIP/france-gateway,60,tr) or exten = _8001133[1-57-9]XX.,1,Dial(SIP/france-gateway,60,tr) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Special Problem in Australia ??
On Sun, 26 Dec 2004 17:04:01 +1100, Eric Bishop [EMAIL PROTECTED] wrote: Just out of interest, what BRI card are you with Asterisk? AVM Fritz!Card PCI. -Shaun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
