[Asterisk-Users] Review Outgoing VM Messages
Hey All, I had a user ask how to go in and listen to her current outgoing messages. I must confess, I can't figure out how to. Any ideas? ~c ___ 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] Any useful results?
Josiah Bryan, Any useful results from your number of installed systems survey? If so, could you email them to me off-list? Thanks, Chris ___ 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] Number of production asterisk systems
Hey Guys, I lost a deal to another vendor because he could point to a number of installations of the product he was selling in our area and nationally, even though _he_ didn't implement them directly. Very frustrating, but I don't imagine uncommon as we compete with other more recognizable solutions. What I am trying to do is track down a rough idea of how many Asterisk systems are in production right now. Ideally as this information was gathered it could be sorted by country, state, industry, etc. Does anyone have any information, or any idea of where to start? Any of the Digium guys on this list know if Digium attempts to track this sort of information? I would be willing to donate time to help compile and correlate the information if anyone has any idea where to start. In the end, I think it would benefit the entire community. I considered posting this to the -biz list but in the end decided that since I was not look for or offering services, goods, etc. that the -user list was a better place. I apologize if you disagree. I know this list is bursting at the seams already. ~chris ___ 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] RE: Ebay listing selling Asterisk @ Home (Blah Blah)
Message: 14 Date: Mon, 11 Apr 2005 17:35:05 -0400 From: dean collins [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Lol, just posted a question to the list that should keep away any bidders. [Christopher Jacob] Why? Is there some reason this person shouldn't make a living selling Asterisk / AMP / FOP etc??? In fact, he is at least fessing up to that fact that it is Asterisk AND open source. While he of course has to include all source (or provide access to it) he doesn't have to advertise the fact that it is Asterisk. Thankfully, people all over the world are selling Asterisk and Asterisk related services. It's what gives the product a foot hold. It's what finances digium. Do you think that the guy that developed AMP did it without intending to make some cash off of it? The released it as OSS, (which is awesome) but of course they are going to continue to sell it. Freak. ~c ___ 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 as test equipment
Hey All, Would it be possible to put a T1 card in an asterisk box and use it to simulate a PRI from the CO? As I build Asterisk boxes for customers I would like to test the installation before getting on site. If this will not work, can anyone share any other ideas for building (or buying) asterisk test equipment. Thanks, Chris ___ 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] PRI Test Equipment
Hey All, I am looking for some advice on setting up a test lab for Asterisk systems that I plan to install at client sites. Most clients will be using PRI. Does anyone have any suggestions on equipping a lab to be able to test Asterisk servers with T1 cards and SIP phones? I have seen some test equipment around the internet, but I'm not sure what the best is. I have also thought about sticking a T1 card in an asterisk box and using a crossover cable to connect the two. Anyone see a reason why this wouldn't work. Home grown, or purchased test equipment. Which is the best way to go? Thanks, Chris ___ 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: Polycom vs. Cisco IP Phones
Oddly enough, I am in the middle of dealing with both a Cisco phone problem and a Polycom phone problem. The Polycom problem I caused myself (oops during a flash) and the Cisco problem came out of nowhere. Polycom During a flash upgrade I lost power (tripped over the cord) and the phone was DEAD. I called polycom, gave them the serial number and they cross shipped a brand new phone to me the next day. I was amazed. With the chatter on this list I was gearing up for a fight. Cisco One of my users came to me to report that Cisco 7960 had gone belly up. These phones always seemed pretty hearty to me, so I was surprised. I checked it out and sure enough not only was it dead, but it would eat any switch port I plugged it into. (seems it was sending power down the Ethernet cable, which is not a good thing) I called Cisco and they said, well you didn't buy it from us so contact the reseller. I contacted the reseller (which is a prominent member of the asterisk community and sells a _bunch_ of asterisk related gear. The first response via email was It is very rare to have a problem with one of these phones. So I email back and said, well it's only a few (less than 6) months old, can I RMA it for a replacement No response So I called, and got someone in Customer Service. He said, oh well we don't buy those phones from Cisco so they wont deal with us either. I will have to contact our supplier and get back to you. Fine, I thought Well a week later I had not received a call. So I called back today. He apologized and promised that he would have an answer for me soon. Now I'm not throwing anyone under the bus here. They are trying to make things right. I am just using my experience as an example. So, the moral of this story While Polycom may not offer configuration type support for asterisk, they stand by their hardware. With Cisco you have to shop around to find a decent deal, and who know how you're going to get support. I don't mind using this list / the wiki / google / etc. for configuration type questions. After all I don't expect Super Micro to help me get FC running on one of their motherboards... Just my thoughts ~c -- Message: 6 Date: Thu, 17 Mar 2005 13:13:14 -0500 From: Noah Miller [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Polycom vs. Cisco IP Phones To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII; format=flowed Not sure the rquirements for your receptionist. I have found that the IP600 does have most everything required to function properly. If you do have an office without DID and a lot of traffic then you may want to look at the tools to display status on her computer. I do have a Snom inhouse for testing when I get a chance. Their quality however is not as good as either Cisco or Polycom. My experience is that the Cisco and Polycom phones are both about in terms audio quality and useability. Neither one does exactly what I'd expect with respect to multiple lines. They both take a little extra setup in this regard, but you can read the wiki for that stuff. Snoms do exactly what I'd expect for a multiple line phone, are very easy to setup, but the audio quality and usability do not compare favorably with either Cisco or Polycom. Between Cisco and Polycom, I went with Polycom just because of cost. The Polycom units are MUCH less expensive (since this is not the biz list, ask me privately about my reseller that is cheaper than others you'll find). On the other hand, Polycom VoIP phones are NOT supported by the company. The only way I've gotten support for them is from this list. Cisco definitely supports all their products quite nicely (for a fee). ___ 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] 7960 Dies when network cable connected
Hey All, I have a user whose Cisco 7960 died the other day. When you disconnect the Ethernet cable the phone boots (as far as it can w/o network connectivity) but as soon as you plug in the CAT5 it goes dead. (no lights, no sound, no display, nothing) I tried to contact Cisco but they say I have to contact the vendor. I have sent an email to the vendor (www.thevoipconnection.com) and I am waiting to hear back from them. I have asked Google in every possible way I can thing of and I have come up empty handed. Anyone have any ideas? Thanks, ~c ___ 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: Asterisk-Users Digest, Vol 7, Issue 93
Date: Mon, 07 Feb 2005 02:22:07 -0800 From: George Pajari [EMAIL PROTECTED] Subject: [Asterisk-Users] Remote MWI via IAX? To: Asterisk Users Mailing List - Non-Commercial Discussion. asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed We have a couple of Asterisk boxes with one being the main system with everyone's voicemail and the other a slave used merely to link a couple of remote phones to the main system using IAX. How can one propagate message waiting indication from the main system to the remote phones? g. I have this same problem and after asking around for awhile I ended writing a TCL script that connected to the remote asterisk server using the Manager API and watching for voicemail. When it catches one it just creates a blank file called MSG0001.txt (I think that's it, but I'm not in front of the machine right now) on the local asterisk box in the appropriate /var/spool/asterisk directory. When the user checks the mail it removes it. Works pretty well. Email me if you want a copy of what I've got. ~c ___ 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] MySQL Realtime Driver
Can someone shed some light on this? It sounds like exactly what I am looking for. Does it handle extensions.conf or just sip/iax/voicemail? (not that to say that _just_ those things would be cool) I have googled for some more information, but so far the only thing I can find is in the bug tracker and perhaps I'm missing something, but I don't get a full explanation. Any insight would be greatly appreciated. ~c ___ 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] retrieve_extensions_from_mysql.pl
Hey All, I want to use retrieve_extensions_from_mysql.pl to build my extensions.conf file from MySQL. Everything seems to be self explanatory except... 1. STATIC= and WRITEPROTECT= (How do I store these in the DB) 2. include = (how to I store / output these? exten = seems to be hardcoded. Any help would be appreciated!!! Thanks, ~c ___ 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] No version string
Hey All, I use cvsup to update my Asterisk servers. I have the following in the sup file... *default host=cvs.digium.com *default base=/usr/src *default release=cvs tag=v1-0 *default delete use-rel-suffix asterisk ;libpri ;zaptel asterisk-addons asterisk-sounds After it downloads the files, I do a make clean make make install When I connect to the console on one machine I get... Connected to Asterisk CVS-v1-0-10/15/04-14:48:10 currently running on bell ( When I connect on the other I get Connected to Asterisk currently running on bell Any ideas why I don't get the correct versions? ~chris ___ 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] (no subject)
Hey All, I would like to start moving all my .conf files into a database backend. Before I go reinventing the wheel, I wanted to check on what is being worked on. I don't think the Wiki has been updated on this subject (which I will be happy to do one I am armed with the information) Any tips, pointers, etc. would be greatly appreciated. ~chris ___ 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] conf from database
Sorry for replying to my own post, but I just noticed my envelope lost it's subject damn outlook. ~c -- Message: 7 Date: Tue, 30 Nov 2004 15:39:32 -0500 From: Christopher Jacob [EMAIL PROTECTED] Subject: [Asterisk-Users] (no subject) To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hey All, I would like to start moving all my .conf files into a database backend. Before I go reinventing the wheel, I wanted to check on what is being worked on. I don't think the Wiki has been updated on this subject (which I will be happy to do one I am armed with the information) Any tips, pointers, etc. would be greatly appreciated. ~chris ___ 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] SysMaster and GPL Violation (lets think before we jump)
Hey All, Isn't it possible that part of the commercial licenses that is offered is that you (the buyer) are not required to advertise, disclose, or even admit that your products offerings are based on an open source project? What other reason would one have for buying a commercial license for an OS piece of software? ~c ___ 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] Remote MWI (I know it's possible)
Hey Folks, I am trying to light a MWI located on a remote SIP phone. In other words, the phones register to one server but the voicemail app lives on a different one. I am guessing it has something to do with passing a user command in the voicemail.conf file. Of course I would also need to clear it... PLEASE let me know if anyone has any ideas. Thanks, Chris ___ 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] Remote Voicemail
Hey all,
I have a bit of a conundrum I need some help with...
I have two servers. local and remote...
The voicemail app lives on remote.
When a local sip dials another local sip that is not available, I need to go
to the remote voicemail app.
Right now I send it to extension 4600 (which is on remote) and use the
switch command to share the dial plan.
exten = 4600,1,wait(1)
exten = 4600,2,VoiceMailMain(s${CALLERIDNUM})
[macro-vmessage]
exten = s,1,Wait(1)
exten = s,2,VoiceMail(u${ARG1})
exten = s,3,Playback(goodbye)
exten = s,4,Hangup
I don't think the CALLERIDNUM is the right variable to pass.
Also, I'm not sure if this is the best solution.
I had hoped that the switch command would allow me to call macros on the
other server, but it doesn't seem to.
Any ideas, help, criticism, etc. would be appreciated.
~c
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[Asterisk-Users] RE: [OT] Sparco Office Supplies... (yeah right)
Come on, this guy accomplished exactly what he wanted to... He got everyone going to Google and searching for Sparco... Don't buy in... just press delete and move on. ~c I don't want to even guess what kinda response this is gonna get... did you even bother using Google? The first result is a store that sells sparco office supplies... perhaps badgering them could turn up a lead? ___ 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: Re: [Asterisk-Users] Digium TheVoice recordings' sound
Email her!!! She is awesome and will of course fix them for you. She doesn't work for digium. Most of the sound files that are included with Asterisk were paid for by various companies and released under GPL so that she would still be able to make a living, and the entire community would benefit. Again, email her... she will make it right, ~c -- Message: 6 Date: Sun, 24 Oct 2004 12:49:46 -0500 From: Kristian Kielhofner [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Digium TheVoice recordings' sound terrible To: Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed Benjamin on Asterisk Mailing Lists wrote: On Sun, 24 Oct 2004 09:27:49 -0500 (CDT), Joe Greco [EMAIL PROTECTED] wrote: http://www.theivrvoice.com/ would seem to imply otherwise. I'd be a bit surprised if any company had enough work to keep her employed full-time, so the works at Digium line sounds a bit fishy to me. I think when he wrote 'She does work at Digium' it was meant in the sense of She is doing work at Digium', or 'She does (some) work for Digium ;-) rgds benjk I would lean towards she does some work for Digium. Did you check out her webpage? One, she lives in Canada, so she certainly does not work at Digium in the physical sense. Two, her client list leads me to believe that Digium is probably one of her smaller clients. Did you try to contact her directly? She seems to imply on her site that customer satisfaction is pretty important to her. Maybe she will fix them for you. You did after all pay for these files, right? ;) She must have a fairly good relationship with Digium, however, because she does have the official title of Asterisk Diva, and she was at Astricon. -- Kristian Kielhofner ___ 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] Multiple Accounts on a Softphone
Hey All, I am looking for a SIP soft phone to deploy at my company. I have been messing around with X-Lite and it looked like the winner, but I have found a problem. I need to be able to set up two accounts on the phone and have the user EASILY toggle between them. You see, I have two asterisk servers. One for internal users (people in the office) and one for external users (people home, customer site, etc.) The problem with xLite is it always seems to try to register the default. If you are out of the office, this is not possible as that box is on the internal network only. Anyone have any suggestions? Thanks Chris ___ 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] RE: answer on # key?
I was asking about this about a week ago. What I found out is that the # option is in the ZAP channel not the dial() command. Ie. Dial(Zap/g1c/5551212) It does work as advertised, but in my mind has some limitations. It sits silently waiting for user input. There is a bug filled, but I don't know the status. In my mind you should be able to play a file. (Press # to accept this call) Others have suggested that it be moved to the dial() command so that it could be used across all channels. I don't know if this is possible. Hope this helps... ~chris Message: 9 Date: Thu, 21 Oct 2004 12:47:15 -0500 From: Matthew Simpson [EMAIL PROTECTED] Subject: [Asterisk-Users] answer on # key? To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; format=flowed; charset=iso-8859-1; reply-type=original I thought I read somewhere on the Wiki that one could give Dial() an argument that would first dial the extension, but not bridge the connection until the called party hit the # key. It must have been during one of those late night coding sessions because now I can't find anything to do with that other than options to allow hangup of the call by hitting *. Does such an option exist? If not, is anyone using a Macro to do that? I have a system that attempts to do a Dial out to a cell phone number with a 15 second timer as a find me type of application. If the cell phone is off or out of range, the 15 seconds of ring time isn't reached and the caller gets connected to the cell phone's voicemail instead of the Asterisk voicemail like I want. Having the # to connect option would fix this problem. ___ 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] Where to buy POLYCOM phones?
Is it like a game?? Let's see who can be an ass first!!! Someone asked a question that I can refer them to google for!! I better mash that reply button so I can be the cool guy!!! It's ridiculous to point this user to Google for this question unless you are pointing him towards a previous thread on the mailing list in which case you should say so... Google - Good source for Polycom site:lists.digium.com Otherwise, what this person is asking for is perfectly reasonable. He said a good place to find polycom phones, which indicates that he wants some feedback from the user community. (which if I'm not mistaken is what this list is for) If he had said Where can I get the best possible price for polycom phones? it would have been a different story as he can easily find that info on Google. But no, he asked for the advice of the community... ~c Message: 13 Date: Mon, 18 Oct 2004 12:52:22 -0500 (CDT) From: Steve Maroney [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Where to buy POLYCOM phones? To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII Google.com Thank you, Steve Maroney On Mon, 18 Oct 2004, Jonathan Miller wrote: Hi all, I'm trying to put together a list of gear w/prices to implement an asterisk system. Does anyone know a good place to buy polycom phones? Their website isn't much help. Specifically looking for IP500 and IP600 phones. Thanks again! Sincerely, Jonathan Miller ACCS.net ___ 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] Zap Channel wait for #
Message: 6 Date: Fri, 15 Oct 2004 08:29:01 -0700 From: Chad Scott [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Zap Channel wait for # To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII; format=flowed Does the option A(filename) not work for you? The A(filename) option will not fire until after the called party presses #. When they first answer the call they only get silence... I have a find me routine that I use but I forward the call with the original caller ID. There are two reasons why I want to use this option... One is to let the called party know that this call is coming via asterisk and the other is to prevent voicemail from grabbing the call before I have a chance to pull it back and move on to the next step. ~c ___ 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] RE: Asterisk-Users Digest, Vol 3, Issue 185
I agree with the DND setting on the phone. This would be my ideal solution. The only problem is, if the user forgets to set it and we ring X number of times, I need to set it programmatically. Do you know of anyway I can do this? ~c Message: 9 Date: Thu, 14 Oct 2004 08:36:14 -0500 From: Eric Wieling [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] DND on SIP To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed Joseph wrote: On Thu, 2004-10-14 at 07:59, Kevin Walsh wrote: Paul Crick [EMAIL PROTECTED] wrote: On the Cisco 7960, I prefer to use the built-in DND facility. Switch it on with the Settings-6-Yes-Save-Back sequence, which is easy to remember once you've done it a few times. To switch it off, simply press the DBD soft-key that should have been present in the first place. The beauty of this method is the fact that the phone will tell you when it's in DND mode. Other methods, involving AstDB etc., will not. I imagine that something could be done to change the dial tone, or something, when an AstDB-implemented DND mode is in use, but this would involve more work. The Cisco always displays a CFwdALL button, and tells you when forwarding is in effect, so perhaps something could be done with that - if being reminded that you're in DND mode is important to you. ___ 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] ODBCexec
Hey All, I have compiled and installed Robert Hanzliks App_ODBCexec and everything seems to load fine. I can use ODBCquery with out problem, but when I call ODBCexec I get an error... [EMAIL PROTECTED] application 'ODBCexec( Which seems odd because when I reload asterisk I see ^@ == Registered application 'ODBCquery' ^@ == Registered application 'ODBCexec' Anyone have any ideas? Thanks, Chris ___ 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] ODBCexec -Fixed-
Hey all, I fixed this if anyone cares to know Like a good little SQLer I ended my query with a ; ODBCexec(select * from somewhere;) Dropping the ; solved the problem. ~c Message: 10 Date: Thu, 14 Oct 2004 17:35:37 -0400 From: Christopher Jacob [EMAIL PROTECTED] Subject: [Asterisk-Users] ODBCexec To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hey All, I have compiled and installed Robert Hanzliks App_ODBCexec and everything seems to load fine. I can use ODBCquery with out problem, but when I call ODBCexec I get an error... [EMAIL PROTECTED] application 'ODBCexec( Which seems odd because when I reload asterisk I see ^@ == Registered application 'ODBCquery' ^@ == Registered application 'ODBCexec' Anyone have any ideas? Thanks, Chris ___ 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] Zap Channel wait for #
Hey all, I am using the c option when transferring to a zap channel to wait for a # before connecting the calling party. It works as advertised, however I would like to play a prompt to the called party. (ie Please press # to accept this call) http://bugs.digium.com/bug_view_page.php?bug_id=0002356 seems to be related. Anyone know what the status is? Or of there is a workaround of some sort? Thanks, Chris ___ 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] Asterisk VIA SSH Tunnels
Thanks for the response... Of course you can SSH in to a machine and run the Asterisk CL. That is not what I am asking about. Specifically I am asking about tunneling. (ie establish an SSH session between my machine and the server, initiating a tunnel on the SIP/IAX ports, and connecting a client ((x-ten or the like)) to the server using localhost as the server address) I know there is a ton of information on Google about SSH Tunnels, and I know that this is theoretically possible, what I was specifically asking for was user experience, not a how do I? I am all about an optimal signal / noise ratio on this list, but just because a topic was discussed once or twice in the past doesn't mean it can't ever be brought up again. As this software evolves, things are bound to change and necessitate revisiting a subject. Again, thanks for the response! Anyone have any experiences (good or bad) trying to accomplish this? Thanks, Chris -- Message: 13 Date: Tue, 12 Oct 2004 23:05:42 -0400 From: Andrew Thompson [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk VIA SSH Tunnels To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Christopher Jacob wrote: Anyone ever set up Asterisk to use SSH Tunneling? Anyone know the pros cons? Asterisk has a command line interface that can be called from probably any shell. I ssh into my Linux box that runs asterisk then tweak my settings/run asterisk -r with no special configuration other than actually turning on and configuring the sshd, which should be done anyway. Are you sure you mean ssh? Could you possibly mean VPN(in all it's varieties)? If you want to know about securing the voip traffic, remove ssh from my previous statement and try these keywords: site:Linux.digium.com ipsec site:Linux.digium.com vpn Sugar to taste... (ie, add any other keywords that you think are helpful) -- Andrew Thompson http://aktzero.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] DND on SIP
Hey All, I have a 2 tier Asterisk system that looks like this... PRI - Remote * - IAX - Local * - Local SIP Users | Remote Sip Users If someone calls a DID we ring (at the same time) The users Remote SIP (SIP/1234) The users Remote SIP (IAX/1234) And a cell phone (ZAP/3015551234) The above works fine, but I have been trying to do some tweaks that I can't seem to get... 90% of our SIP users are soft phone, so the above is fine for them. However, we have a group of users who are Cisco 7960G when local and softphone when remote. In that scenario, the Cisco is always present and logged in, so always rings. Here is what I would like to do, all ideas and comments are welcome... When the Cisco user is in the office he/she can hit a button on the 7960 that toggles in/out. (It has been requested that this be a one touch option, or in other words not buried 2 or 3 levels deep) When a DID is dialed, we check the status of the toggle state. If they are in we ring the phone for 20 seconds, if they don't answer we continue the dial plan. (remote SIP followed by cell) at the same time, we toggle them to away so it doesn't happen again the next time. If they are out we just jump to the next step (remote sip followed by cell) Am I missing something really simple? I have been and will continue to wander google and voip-info, but any experience anyone has is greatly appreciated. Thanks, Chris ___ 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] Asterisk VIA SSH Tunnels
Anyone ever set up Asterisk to use SSH Tunneling? Anyone know the pros cons? Thanks, Chris ___ 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
Ok, plenty of Cheap Shots all the way around... Can we call this a draw? I believe what Paul was saying is IF the attitude from the list is don't buy digium hardware don't come around here for help, than that is not in keeping with the general OS community. I happen to agree. I buy Digium hardware because I love the idea of sell the hardware open source the code. (precisely the same reason I own a Squeeze Box) Not to mention great quality, great services, etc. etc. However, you have to understand $10 vs. $100 for proof of concept. Let's work together to make OS the norm and not the exception. ~c -- Message: 5 Date: Sun, 10 Oct 2004 13:49:39 -0500 From: Brian Capouch [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Intel Modem vs Digium Cards To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed Wolf Paul wrote: However, those of us not working with hefty corporate budgets may not have the option of spending $100 for a test machine when there's a more cost effective option available. . . . . . If that warrants don't come asking for support then you guys are not much of a community but a sales machine for Digium. Cheap shot. Digium does Asterisk FOR FREE. 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. 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. Stir into that mix a first-timer who is undercutting the profit model that enables Digium to offer us this wonderful software, and then sprinkle your obnoxious insult to the community on top, and you're going to find that people (correctly) tell you to go away and solve your own problems. From my perspective your primary problem isn't hardware; its your attitude. B. *** ___ 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
Cisco Downloads -- was -- Re: [Asterisk-Users] Cisco 7960 and Asterisk...not working...
I just had to deal with this yesterday. I called Cisco and they gave me a part number for the support contract. I looked around and it was $90... I posted back to this list and was happy when someone gave me the correct part number, which at CDW was $10... Not too bad. Although I can't believe Cisco waists time with $10 service contracts. At that point just make the damn thing free. Anyway, I don't have the email in front of me but google the list from yesterday (Friday) and I'm sure you will come across it. ~c Message: 10 Date: Sat, 25 Sep 2004 18:45:44 + From: C Wegrzyn [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 and Asterisk...not working... To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Is there a place to get the software load for the Cisco phone without having a support contract? Buying the phone was costly enough, but now needing to pay for the software to fix it is really poor! Chuck Wegrzyn Chad Brown wrote: Chuck, The first thing I would do is to upgrade the load to version 6 or higher. I'm running the latest...version 7.2. (I'm very happy with it) Are you using TFTP to load the configuration or manually configuring the 7960? I know it's a pain to setup TFTP just for a quick test. However, it's well worth it. If you have a CCO account you can find the latest load and config files here: http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960 After getting the infrastructure in place the following link was all I needed to get my 7960 phones working properly: http://www.voip-info.org/wiki-Asterisk%20phone%20cisco%2079xx However, the 7960 does have some basic error logging. I'm not sitting in front of it right now so I can't tell you the key combinations. Hint: I went from version 3.2 like you to 7.2. However, as an interim step I had to go to 6.0 first. Thanks, Chad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C Wegrzyn Sent: Saturday, September 25, 2004 11:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Cisco 7960 and Asterisk...not working... Hi! I have Asterisk up and running and have KPhone, IAXtel and FWD connected to it and working. I have a Cisco 7960 with version 3 (App. Load ID POS3-2-00) software. I have configured the 7960 correctly, I think; I have set everything - name, shortname, auth.name and display name set to 200. I have set the password to 200. I've set the proxy address/port to 192.168.1.117/5060. I can't seem to get the phone to connect to Asterisk, though Kphone works fine. Does anyone have an idea of what I am doing wrong? TIA, Chuck Wegrzyn ___ 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 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] Cisco SIP Files
I am in the process of ordering a support contract from Cisco for my new 7960 phone, but I would really like to get it up and running. At the risk of being flamed off this list, could someone send me or point me in the direction of the SIP image files I need to change the phone over? Thanks, ~c ___ 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] Cisco Support Agreements
Hey all, I am trying to get the Cisco SIP image loaded on my new 7960. The Wiki and several emails in the list archives say the cost is approx. $8.50 per phone per year. The problem I have is the Cisco is giving me part number CON-SNT-PKG1 which costs $90.00. I believe this covers phone support as well. I don't want the phone support, only the software. Does anyone have the part number for software upgrades? (the $8 one referenced on the Wiki) Thanks, ~chris ___ 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] What type of PRI setup is best
I am having my colo set up 2 PRI's for my new asterisk implementation. They asked the following... ##SNIP## What type (NI2, NTI, 4ESS, or 5ESS) and whether they want to be USER or NETWORK. If the equipment is flexible, NI2, with us as NETWORK is preferred. ##SNIP## We are using a digium quad span T1 card. What is the recommended setup? Thanks, Chris ___ 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] Asterisk and SER
Hi All, Can someone help me clear up some stuff? I am about to implement asterisk for a office of about 20 people. I plan on running SIP phones for everyone. (a mix of Cisco Sets and Xlite soft phones) We will place the Asterisk server at a collocation provider and have in connected to the PSTN via 2 PRIs. (digium card) When customers call our 800 number they will be sent to asterisk. When they enter an extension I want asterisk to check if that SIP users is logged in and if not transfer the call back out over PSTN (to a cell phone) Now, here is where things are a little foggy... I want put a local Asterisk server here in the office so that the SIP users connect to it thereby reducing the chatter across the WAN. I would like to have the two Asterisk servers communicate via IAX. Questions: 1. Does this scenario pass muster? Is my thinking logical or does anyone have a better suggestion? 2. Is this possible? Can the remote Asterisk server check to see if the SIP user is logged in to the local Asterisk server before sending the call across the WAN? 3. Should I be using SER vs. another Asterisk server? The problem I see with this is that it doesn't support IAX. I believe that is the preferred method? Am I right? Thanks for all the help from the OSS community. Great software!!! ~chris Christopher Jacob Eye Street Software Program Manager,14151 Newbrook Drive Partner Products Suite 250 301.305.0991Chantilly, VA 20151 www.eyestreet.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] Beyond T1
All, This may be a stupid question, but here it is... What interface gives the most density? Do I top out at T1's? For instance, 4 t1's to the Digium Quad span t1 card. Is there an interface available for T3 or DS3? Thanks, Chris ___ 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] Transfer and Release of a call out to PSTN
Hi Again All, When using Asterisk with a PRI to the CO is it possible to transfer a call back out and release. In other words, once the call is connected (caller and external 3rd party) Asterisk is removed from the equation thereby freeing the PRI channels. I ask because my scenario is going to require frequent external transfers and I would like to control the PRI costs. Could this be done using SS7? If so, does anyone know if any Asterisk SS7 development is being done? Thanks Chris ___ 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] design check
Hey Guys, I am getting ready to implement Asterisk for my company. We plan on housing an asterisk server at a local termination provider and have another here in the office. The two communicate via IAX. Looking something like... PSTN | PROVIDER | (PRI) Asterisk | (IAX over the Internet) Asterisk | SIP Users The idea here is that the local sip phones can communicate with the local Asterisk server and cut down on network traffic across our corporate internet connection. Also, I would like users who are not in the office (travel, home office, etc.) to be able to connect directly to the Asterisk box at the provider. This will further cut down on local traffic. Finally, when a call comes into the Asterisk box at the provider, I would like that box to check if the requested user is logged in down in the office, and if not don't send the call across the internet. Instead, see is the person is logged in directly. If the user is not present at all, try a cell phone or send to VM. Again, this would limit traffic across our corporate internet connection. Has anyone had any experience with this type of implementation? Does this make sense, or am I making things to complicated? Is this configuration and functionality possible? Any other comments or suggestions that may help me accomplish my goal of providing a telecom infrastructure to a company of 20 software engineers who are mostly in the office but could be anywhere. Thanks a bunch guys. ~chris ___ 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] T100P PCI or PCI-x
Hey all, I have looked around the Digium site to no avail. Can someone tell me of the T100P is pci or pcix? Do I need to look for anything specific when selecting a motherboard? Also, the spec sheet says it will fit in a 2U case, but in the picture it looks like it would be possible to squeeze it into a 1u. Anyone have any experience? Thanks Chris ___ 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] SIP Transfers (Possibly reinvite)
Hey Folks, Is it possible to transfer an incoming call back out without a trombone effect. For instance; Caller dials my broadvoice # -- Asterisk Answers and plays a menu -- the caller selects an option -- asterisk transfers the call to my cell phone via broadvoice and removes itself from the equation so I end up with... Caller -- Broadvoice -- Cell Phone Vs. Caller -- Broadvoice -- Asterisk -- Cell Phone Any ideas on how this could work? I'm thinking it's something to do with reinvite. Thanks Chris ___ 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
