[Asterisk-Users] Blind Transfer / Ring Groups Problem
Greetings Iwas using a Asterisk system runnning version 1.0.7 and have just build a new box with version 1.2.5. I have come across a problem however that did not exist before. Setup: 5 Polycom SIP phones 2 PSTN lines into a TDM400 card Calls come into a call group#200 (2 phones) Then, whoever answers, transfers the call (regular transfer or blind transfer) This worked fine before Problem Call comes into call group Person answers and then then transfers 1) Regular transfer works fine 2) Blind Transfer rings the transferedextension PLUS the call group again. 3 phones ring. If I call another extension and that person does a blind transfer, it also works fine. The only situation is when the originating call is to a call group. It does not matter if the call comes from the PSTN or an internal phone. From debug, it appears that the variables sent to the dialparties.agi (methodology and extension map) are not reset to the one extension being transferred to. Instead, they keep the values from the original call. Anyone run into this and is there a fix. Thanks Martin ___ --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] Web Vmail Question
I install WebVmail today on a Fedora 2 box. I got the cgi script running etc and I get the login prompt. However, when I enter a mailbox and password, ie. 201 and 1234, I always get a message saying the login is incorrect. Any tips out there? Thanks Martin ___ 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] Power Alarm Error - Help
Yes, The card is working fine most of the time. It just gets this message on occasion and then Asterisk shuts down. I debating putting surge suppressors on the PSTN lines. Could this be caused but a voltage issue from the Telco? Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, January 23, 2005 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Power Alarm Error - Help I have been getting the following message in Asterisk and it shuts Asterisk down, needing a reboot. Power alarm on Module 2 I have (1) TDM400P with (2) FXS (2) FXO cards (1) X100P card Any ideas? Since nobody answered, I'll guess something :) Did you plug the power on the TDM400P ? since you have FXS ports, you need to plug it in ___ 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] Power Alarm Error - Help
I have been getting the following message in Asterisk and it shuts Asterisk down, needing a reboot. Power alarm on Module 2 I have (1) TDM400P with (2) FXS (2) FXO cards (1) X100P card Any ideas? Thanks Martin ___ 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] ADSI phones and the Flash Key
I have some Sayson / Aastra 480e ADSI phones. They work great except for one annoying feature. It's not really the phone, it's the ADSI programming feature in Asterisk. I can't figure out how to recreate the factor default flash screen key on the phone. When I create a ADSI script, it loads to the phone no problem. However it wipes out all of the factory default keys. I can't figure out what to put in the script to recreate the flash key. Help anyone? Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Intel Modem vs Digium Cards
Why don't you take this off-line were it belongs Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Greco Sent: Monday, October 11, 2004 9:17 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Intel Modem vs Digium Cards Cheap shot. Digium does Asterisk FOR FREE. No. As with most of us who support free software projects, we support them because it suits our business goals. We don't do it for free. The investment in time, effort, and resources is paid back, frequently in a way which can't directly be translated by accountants, but it is still an investment, and it is expected to pay off. There are massive benefits to having other users in the community contributing towards and extending the development. Some of us don't even actively *advertise* our company's association with the project in question, something which has been mildly nagging at me about the Digium situation. They support themselves, which I hope you agree is a necessary thing, by selling hardware, one instance of which is the low-end X100P. Essentially the X100P is a slightly modified generic voicemodem THAT COMES WITH CUSTOMER SUPPORT. That is, along with its hardware functionality comes the ability to call up and get help if you encounter problems. That seems quite reasonable. This list is intensely active, and the developers and others who provide advice here are necessarily limited in the amount of attention they can devote to (the often repetitive) questions coming from first-timers. That seems quite reasonable as well. There are, of course, many other participants on the lists, and numerous resources which can be used to help solve problems. Stir into that mix a first-timer who is undercutting the profit model that enables Digium to offer us this wonderful software, And don't forget to trivialize the contributions of everyone else while you're doing it, and then sprinkle your obnoxious insult to the community on top, I didn't find it obnoxious or insulting. In fact, I'd have to agree. One of the benefits to the whole free software movement is supposed to be the freedom to make choices (or, if you prefer, the freedom not to be locked in to a vendor). If you're going to jump all over a guy who *wants* to join the community, for not buying your Approved Vendor's Hardware, maybe because he can't afford it or justify the cost, then it is you who are damaging and limiting the growth of the community. I would imagine that Digium made a conscious choice to use an existing generic voicemodem chipset and to make its drivers compatible with generic versions. As a manufacturer, they certainly had the option to obfuscate things at the hardware level - and they didn't. If they truly wanted to discourage people from doing this, why distribute a driver package that recognizes and installs generic devices? I believe Digium recognizes that they are adding significant value to an otherwise-worth-$2.50-in-quantity, and are betting that most people will see value in buying in at a premium. However, it appears to me that they have also chosen to invite people in who, for whatever reason, have not chosen to purchase their hardware. Looking at it from their point of view, that makes *sense*, because if someone invests five bucks at Fry's on a crummy softmodem, puts it in their box, discovers the joys of Asterisk, and then sells other people on the wonders of Asterisk, Digium still stands to profit. The community grows, and being the main supplier of Asterisk-compatible interface cards should remain a profitable business because most commercial installations will want some level of support. So for heaven's sake, don't dump on some guy for buying a generic softmodem so he can play around. Encourage it. Say generic softmodem is better than alienating this guy. and you're going to find that people (correctly) tell you to go away and solve your own problems. Wow, that's a really sucky attitude. I would expect *Digium* to tell him to go away and solve his own problems. However, if the user community does that, then this is one of the suckiest user communities I've run across in the free software world, and I've been doing free software for many years. From my perspective your primary problem isn't hardware; its your attitude. And from mine, it's users with attitudes like yours. As for me? I'm shopping for cheap modem cards. Why? 1) I'm on FreeBSD, so Digium probably won't support that. 2) I realistically expect to go all VoIP, except perhaps for fax, so I don't want to spend a ton on cards that I won't need. 3) I expect to do something like a Sipura 3000 if we retain a single POTS line, or maybe some sort of Cisco with ISDN BRI VIC cards if we keep the BRI's. 4) I don't really think my PPro200 PBX box will survive very well with having to handle the codec work anyways. But I'm open to spending ten bucks to
[Asterisk-Users] Restart Digium Cards
Title: Message Greetings I have a X100p and a TDM400P card in a Redhat 9 server. If Asterisk is not suhtdown properly (reboot without stopping Asterisk or power failure etc), I have to do a modprobe command after the restart before Asterisk works again. Any ideas how to resolve this. Martin
RE: [Asterisk-Users] Restart Digium Cards
Thankyou, Thanyou, Thankyou : Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanwar Ranbir Sandhu Sent: Wednesday, August 18, 2004 2:52 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Restart Digium Cards On Wed, 2004-08-18 at 15:38, Martin Keding wrote: I have a X100p and a TDM400P card in a Redhat 9 server. If Asterisk is not suhtdown properly (reboot without stopping Asterisk or power failure etc), I have to do a modprobe command after the restart before Asterisk works again. Any ideas how to resolve this. That's an easy one, and I am sure it's in the wiki or in the list, somewhere. However if others are wondering, here's the solution: in /etc/rc.modules, type in the modprobe command you use to manually load the modules. That's it! Next time the server reboots, the rc.modules file will be read and the modules will be automatically loaded. HTH, Ranbir -- Ranbir Systems Aligned Inc. www.systemsaligned.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS Number after Update
Whenever I do a Show version, I get CVS-02/10/04. I have updated and recompiled Asterisk a number of times and it comes up clean. Should'nt this show the lastest version number? I have been using CVS download, then make clean;make install. I have also tried make clean; cvs update; make; make install Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RC2 - RH9
Title: Message This may be a stupid question but I have tried compiling RC2 on my RH9 box and everything appeared fine. What version should I be seeing when I type show version though. Martin
[Asterisk-Users] Z110p card linh hang up
This question may have been answered 100 times already but I am new to the list. Sorry. I have an x100p as my main PSTN (Canada). Everything works fine but the x100p takes a very long time to hangup on calls. It takes up to 30 secs, before the card will receive a new call. What is really frustrating is that a caller who hanges up during a please leave message prompt, still generates a message, even if they hang up without leaving a message. Is there a way improving this? Thanks for your help. Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of T. Chan Sent: Sunday, August 01, 2004 3:37 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk scalability? Hi, Scott Thanks for your information. I have worse luck in load testing with asterisk. I have tried both SIP and H323 inbound calls and terminating on PSTN PRIs. I am using a single Xeon 2.8G chip and 512M Ram and in both cases, once it gets more than a T1, call quality starts to degrade with choppiness, and Asterisk becomes very unstable and resets itself like every 5-15 minutes. Can you let me know more about your tests, like which version of Asterisk are you using for the test, and which version of H323 and your computer configuration please, thanks a million TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Saturday, July 31, 2004 2:16 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk scalability? Hi Roy- I've done a lot of load testing with asterisk and TE410P's. My guess, with no transcoding, is that you might be able to handle 8 E1's max on the PSTN side absolute max (ie: 2 TE410P's). This assumes you have a fast processor.If you're using T1's, scale these numbers up accordingly, as there are fewer channels per span. If this answer is lower than you might expect, consider that every byte of data has to pass through the processor. The 410's are capable of bus-mastering, and so are an improvement over the T400P's, but still I think you run into horsepower limitations. Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk Sent: Saturday, July 31, 2004 8:25 AM To: Asterisk Users Subject: [Asterisk-Users] Asterisk scalability? Hi I plan to setup an asterisk box to function as a SIP gateway forwarding lots of calls to/from a backend of several other asterisk boxes, each with a TE410 card for PSTN connectivity. It will only gateway the calls into the PSTN gateways. No transcoding is planned - only plain ALAW. How many concurrent calls would you think this can handle? I'm asked to plan a system that can handle 1000 concurrent calls... thanks for any input regards roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.732 / Virus Database: 486 - Release Date: 7/29/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.732 / Virus Database: 486 - Release Date: 7/29/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] 480i User Feedback With Asterisk 802.1Q?
I just brought in a 480i for testing. It is VERY bare basics! Part of the Web interface still doesn't work (doesn't show you the sip setting) and barely has any other settings available. Also does not have any features for NAT. The only way of programing it currently with TFTP. I still haven't got it to register properly with Asterisk. Released a little to early I think. However, it is a very professional looking phone and once the bugs are out of it, it seems very promising at a good price. Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, August 01, 2004 5:21 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 480i User Feedback With Asterisk 802.1Q? On Sat, 31 Jul 2004, Kevin wrote: Does anyone know if the 480i supports 802.1Q? I don't see any support for it at the moment, but this is a very early firmware, with a bare minimum of features. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Aastra 480e phone ADSI config
THEN CLEAR ; SHOWDISPLAY titles AT 1 NOUPDATE ; SHOWDISPLAY incoming AT 2 NOUPDATE SHOWDISPLAY callname AT 3 NOUPDATE SHOWDISPLAY callnum AT 4 ENDIF IFEVENT RING THEN CLEAR SHOWDISPLAY titles AT 1 NOUPDATE SHOWDISPLAY incoming AT 2 ENDIF IFEVENT ENDOFRING THEN SHOWDISPLAY missedcall AT 2 CLEAR SHOWDISPLAY titles AT 1 SHOWKEYS vmail_OH ENDIF IFEVENT TIMER THEN CLEAR SHOWDISPLAY empty AT 4 ENDIF ENDSUB SUB offHook IS IFEVENT FARRING THEN CLEAR SHOWDISPLAY titles AT 1 NOUPDATE SHOWDISPLAY ringing AT 2 NOUPDATE SHOWDISPLAY callname at 3 NOUPDATE SHOWDISPLAY callnum at 4 ENDIF IFEVENT FARANSWER THEN CLEAR SHOWDISPLAY talkingto AT 2 GOTO stableCall ENDIF IFEVENT BUSY THEN CLEAR SHOWDISPLAY titles AT 1 NOUPDATE SHOWDISPLAY busy AT 2 NOUPDATE SHOWDISPLAY callname at 3 NOUPDATE SHOWDISPLAY callnum at 4 ENDIF IFEVENT REORDER THEN CLEAR SHOWDISPLAY titles AT 1 NOUPDATE SHOWDISPLAY reorder AT 2 NOUPDATE SHOWDISPLAY callname at 3 NOUPDATE SHOWDISPLAY callnum at 4 ENDIF ENDSUB SUB stableCall IS IFEVENT REORDER THEN SHOWDISPLAY callended AT 2 ENDIF ENDSUB ; - ; End Asterisk default ADSI script ; -; 3. I only had to tune the SENDDTMF 8500 values to properly send it to the right voicemain extention 4. Added the following to my /etc/asterisk/extensions.conf file in a local only context so that the phone could only be programmed locally: [adsi-program] exten = 9666,1,Authenticate(1234) exten = 9666,2,ADSIProg(asterisk.adsi) exten = 9666,3,Hangup 5. Called extension 9666 from the 480e. It asks for my password and then I am off to the races. Good luck! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Keding Sent: Wednesday, July 28, 2004 5:44 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Aastra 480e phone ADSI config Greetings Does anyone have a ADSI config file for an Astra (Sayson) 480e phone. I am using the sample asterisk.adsi file but if anyone already has a modified working file that they would like to share, could you let me know. Thanks Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Astricon Recordings?
Just a died question. Will all of the sessions be recorded and made available? Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Thursday, July 29, 2004 10:34 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER * Scott Stingel wrote: Hi Olle- I wonder of you could please post the most recent agenda for each day, even if it's not finalized. Some of us can't attend the whole conference, and so need to pick the best days/times to come. (I'm scheduling a trip, and a stop at astricon could be on the way there) The most recent agenda is always on the web site - you have all the details there and I update as soon as I know there's a change. http://www.astricon.net /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice mail problem
I am having a problem with getting voice mails, even when the caller hangs up before getting to the recording prompt. If I call my number, even if I hang up the second I get the I'm not in recording, it still generates a voicemail. Is there a way around this? Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] HELP! With Postresql
I installed Postresql and then recompiled Asterisk. I understood that Asterisk would see Postresql on the recompile and add it. Is there a way of checking? Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen R. Darragh Sent: Tuesday, July 27, 2004 10:54 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] HELP! With Postresql Have you actually compiled the pgsql CDR module in to Asterisk? On Wed, 2004-07-28 at 09:43, Martin Keding wrote: I am having some real problems with getting CDR records to go to a Postresql database. I think I have followed every post and instruction available and Asterisk still happily writes to a text file. Postresql is installed and working on a Redhat 9.0 box, the same one as Asterisk. I have created the CDR table in a database called Asterisk. Conf files etc are set. I even recompiled Asterisk. Any pointers would be greatly appreciated. Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Stephen Darragh Technical Director Informed Technology Ph: +61 8 9380 4244 Fax: +61 8 9380 4354 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Aastra 480e phone ADSI config
Title: Message Greetings Does anyone have a ADSI config file for an Astra (Sayson) 480e phone.I am using the sample asterisk.adsi file but if anyone already has a modified working file that they would like to share, could you let me know. Thanks Martin
RE: [Asterisk-Users] HELP! With Postresql
Thanks for your help The cdr_pgsql.so was not there. Do I change the mods line to MODS=cdr_csv.so; cdr_pgsql.so Ie. Do I add a semi-colon or not. Thanks Martin Hi - Check /usr/lib/modules to see if cdr_pgsql.so is in there. If not, edit the Makefile in the asterisk/cdr directory and add cdr_pgsql.so to the MODS= line near the top of the file and then rebuild. You may need to edit some files if your postgresql headers aren't where the source expects. Is asterisk going to get an autoconf script any time soon? On Wed, 2004-07-28 at 22:52, Martin Keding wrote: I installed Postresql and then recompiled Asterisk. I understood that Asterisk would see Postresql on the recompile and add it. Is there a way of checking? Martin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen R. Darragh Sent: Tuesday, July 27, 2004 10:54 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] HELP! With Postresql Have you actually compiled the pgsql CDR module in to Asterisk? On Wed, 2004-07-28 at 09:43, Martin Keding wrote: I am having some real problems with getting CDR records to go to a Postresql database. I think I have followed every post and instruction available and Asterisk still happily writes to a text file. Postresql is installed and working on a Redhat 9.0 box, the same one as Asterisk. I have created the CDR table in a database called Asterisk. Conf files etc are set. I even recompiled Asterisk. Any pointers would be greatly appreciated. Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Stephen Darragh Technical Director Informed Technology Ph: +61 8 9380 4244 Fax: +61 8 9380 4354 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP! With Postresql
I am having some real problems with getting CDR records to go to a Postresql database. I think I have followed every post and instruction available and Asterisk still happily writes to a text file. Postresql is installed and working on a Redhat 9.0 box, the same one as Asterisk. I have created the CDR table in a database called Asterisk. Conf files etc are set. I even recompiled Asterisk. Any pointers would be greatly appreciated. Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multi companies
I am fairly new to Asterisk and I want to do some testing with multi-companies on the same box. I have two inbound lines and I basically want one to trigger auto-att. for company 1, the other line to trigger auto-attend for company 2. Could somebody point me to a sample conf. or documentation. Thanks Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
