Re: [asterisk-users] Is there a possiblity to check in the dialplan whether a SIP user is registred?
ChanAvail() [EMAIL PROTECTED] wrote: Hello everybody, Is there a possiblity to check in the dialplan whether a SIP user is registred? Something like : exten => user1,1,GotoIf(isRegistred(user1)? context1, context2, 1) Thanx, Kalle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!
The polycom lets you do either attended or unattended transfers. If you want an unattended transfer, you press the 'blind' soft key. It's been a few months since I've looked at this, so a bit fuzzy on the details. Jason Adams wrote: Isn’t that the function of an attended transfer? User3 hears User1 to see if they want to take the call or not. User1 should then hit the transfer key again to finalize the call. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Jim Suber *Sent:* Thursday, May 03, 2007 12:54 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Asterisk-Polycom HEPPP PBX: Asterisk 1.4 Phones: PSTN phone connected to TDM400 X-Ten Lite Polycom 430 Scenario Polycom 430 = User1 User2 calls User1(Polycom 430) asks to be transfered to User3 User1 does an attended transfer using the trnsfr button on the polycom User2 is placed in music-on-hold User3s phone rings. (So far so good Right?) User3 picks up the phone to answer User2 only to find that he is talking to User1 User2 is stuck in music-on-hold. FOREVER! The other two phones work exactly as they should using the # key Using the # key on the Polycom only allow the dialing of 1 number before Alice announces That there is no such extension. HELP Thanks in advance Jim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large dial plans and variables
Andreas Sikkema wrote: You're so right! I thought about having just a catchall _. extension in the dialplan and doing everything else in a "real" language via AGI - PHP, Perl, ... whichever you like. It would make the programming part much easier as the scope of variables is just as you expect it to be. Well, they're called macro's for a reason You guys are proposing adding functions or procedures. My first step in any macro would be to copy incoming variables, be it arguments or even asterisk defined stuff to local variables. But that is just me and my coding convention. I guess we are. I propose we add functions or procedures! Until that time though, it seems best practices are to prefix every single variable in macros, including copying ARG parameters to variables, with the name of the function, to avoid stepping on yourself. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Large dial plans and variables
Philipp Kempgen wrote:
Doug Garstang wrote:
I have a large dial plan here with over 3000 lines, and several dozen
macros. As it grew, it became apparent that there was some problems.
1. When you pass arguments to a macro in the form of $ARG1, $ARG2 etc,
if that macro calls another macro, and passes arguments like this as
well, you lose the original values.
2. When the macro's 'return' some value, it has to set a channel
variable. If your not careful, this quickly becomes a mess. Some
standard mechanism is needed.
In fact, the over all problem is that all channel variables are global
within that channel. Macro's don't have local variables and it makes
programming large dial plans problematic. Even things like loop counters
can get trashed when your inside of a loop, and you jump somewhere else
and modify that loop variable, and jump back.
Anyone got any tips on how to manage this? It would be awesome if AEL2
could address this somehow...
A possible "solution" would be to prefix all variables used
within a macro with the name of the macro but you really feel
it's a workaround.
macro do_something( do_something_foo, do_something_bar ) {
Set(do_something_i=0);
//...
}
But how nice is ${do_something_i} ?
Regards,
Philipp
Philipp, that's what I've been doing. For readability sake, I normally
have macro's with long descriptive names, like macro-VMRetrCheckGeneral
for example. That;s ok until you start suffixing variable names to the
end. You might end up with:
VMRetrCheckGeneral_new
VMRetrCheckGeneral_old
VMRetrCheckGeneral_i
and so on. I guess it's better than nothing
I feel that the lack of local macro variables is a big problem for the
dial plan. Until it's fixed it will hamper the development of large dial
plans.
___
--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] Large dial plans and variables
I have a large dial plan here with over 3000 lines, and several dozen macros. As it grew, it became apparent that there was some problems. 1. When you pass arguments to a macro in the form of $ARG1, $ARG2 etc, if that macro calls another macro, and passes arguments like this as well, you lose the original values. 2. When the macro's 'return' some value, it has to set a channel variable. If your not careful, this quickly becomes a mess. Some standard mechanism is needed. In fact, the over all problem is that all channel variables are global within that channel. Macro's don't have local variables and it makes programming large dial plans problematic. Even things like loop counters can get trashed when your inside of a loop, and you jump somewhere else and modify that loop variable, and jump back. Anyone got any tips on how to manage this? It would be awesome if AEL2 could address this somehow... Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Capture Asterisk traffic
I remember an app called 'vomit' that could allegedly reconstruct audio files from tcpdump pcap files. Salvatore Giudice wrote: I think you want: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp dst portrange 5060-65534 dst port port True if the packet is ip/tcp, ip/udp, ip6/tcp or ip6/udp and has a destination port value of port. The port can be a number or a name used in /etc/services (see tcp(4P) and udp(4P)). If a name is used, both the port number and protocol are checked. If a number or ambiguous name is used, only the port number is checked (e.g., dst port 513 will print both tcp/login traffic and udp/who traffic, and port domain will print both tcp/domain and udp/domain traffic). src port port True if the packet has a source port value of port. port port True if either the source or destination port of the packet is port. dst portrange port1-port2 True if the packet is ip/tcp, ip/udp, ip6/tcp or ip6/udp and has a destination port value between port1 and port2. port1 and port2 are interpreted in the same fashion as the port parameter for port. src portrange port1-port2 True if the packet has a source port value between port1 and port2. portrange port1-port2 True if either the source or destination port of the packet is between port1 and port2. Any of the above port or port range expressions can be prepended with the keywords, tcp or udp, as in: -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of CSB Sent: Tuesday, May 01, 2007 1:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] OT: Capture Asterisk traffic I want to capture all my Asterisk traffic (including RTP) and then analyse it. My plan was to use tcpdump and then analyse with Wireshark. The following works: tcpdump -i eth0 -s 0 -w /tmp/tcpdump.1 But I want to be a bit more selective: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp and dst port >= 5060 This doesn't capture the RTP traffic. Could anyone advise what I'm doing wrong or suggest a better way? Thanks Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MYSQL application in dial plan
Well, you should be able to leave it open. However, I don't know what would happen if MySQL times out and disconnects the connection because it considers it stale. I don't know if you can check that error and reconnect. Yehavi Bourvine +972-8-9489444 wrote: Hello, I would like to implement a few decision making process inside the dialplan using information stored in MySQL (like LCR, etc.). I see the MYSQL() application, but as far as I understand I have to connect to the database each time I want to query it; this seems a CPU eater to me. Is this indeed the case, or can I open it once Asterisk starts and leave it open? Thanks, __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail on Different Server
No, you can get Asterisk and NFS to work fine together. It was in my past job, so I can't remember the exact settings, but there was some magic combination of NFS client mount settings that would cause Asterisk to return immediately, rather than hang, if there was an NFS communications problem. Doug. Anthony Rodgers wrote: It will stall asterisk - ask me how I know.. :-) CP Gordon Henderson wrote: On Tue, 24 Apr 2007, Forrest Beck wrote: > I've heard there are problems using NFS as a storage device.??? I've used NFS for many many years on 100s, maybe 1000s of servers in this time. It's great. "Just works" and does exactly what it says on the tin. I use it at home, for my clients, on my hosted servers, everywhere. (well, almost!) BUT... If the NFS server should go offline for whatever reason then the client systems that are reading/writing the data will stall, and depending on how you've got them setup they will stall hard and not continue until the server returns. I haven't tried it with asterisk yet, so I do not know what will happen to the voicemail system should the NFS server go offline for whatever reason. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk cookbook
What a cool idea! J. Oquendo wrote: http://etel.wiki.oreilly.com/wiki/index.php/Main_Page ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] App Read() kills call when file doesn't exist
I don't know if this is a recent issue... When the read() application is
given a file that does not exist, it aborts the ENTIRE dial plan. That
can't be right. Playback() and Background() don't do this. Couldn't find
a bug in mantis for it...
[Apr 26 17:20:12] VERBOSE[14611] logger.c: -- Executing
[EMAIL PROTECTED]:5] Read("SIP/xx.yy.69.210-009c3f40",
"Key|/usr/local/vm/539/2-greet|1|n|1|1") in new stack
[Apr 26 17:20:12] VERBOSE[14611] logger.c: -- Accepting a maximum of
1 digits.
[Apr 26 17:20:12] WARNING[14611] file.c: File /usr/local/vm/539/2-greet
does not exist in any format
[Apr 26 17:20:12] WARNING[14611] file.c: Unable to open
/usr/local/vm/539/2-greet (format 0x4 (ulaw)): No such file or directory
(hangup routine called here...)
Is this expected behaviour?
Doug.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SP 601 Reboot Issue- Help!
That used to happen to us _ALL_ the time. Sometimes you'd just have to press the 'Directory' key and the phone would instantly reboot. It was very easy to reproduce and Polycom where useless at admitting it might be a problem. It occurred on several phones. Funnily enough, the phone it was most reproducable on was a 601 being used as a Receptionist phone with 3 sidecars... and about 35 buddies being watched. Hmmm! Russ Beaupre wrote: We had a situation where the 601 base went missing and the electrical connection between the side cars and the 601 was broke. Might be worth a look to see if the phone got damaged. -Original Message- From: Jerry Jones <[EMAIL PROTECTED]> To: Asterisk Users Mailing List - Non-Commercial Discussion Date: Tue, 24 Apr 2007 12:27:46 -0500 Subject: Re: [asterisk-users] Polycom SP 601 Reboot Issue- Help! The only reboot issue I have with 1 sidecar is the side car deciding to randonly rebbot, not the phone itself Perhaps upgrading to 2.1 will help? On Apr 24, 2007, at 10:51 AM, J French wrote: > I have a Polycom 601 with 3 expansion modules running 2.0.3. We > have Buddywatch set up on around 42 users on the expansion > modules. We are experiencing reboots on the 601. Today it > happened twice after users paged through the phones. The page > groups have about 23 phones each. There is a third page group > comprising all 46 phones. I'm thinking it may be an issue with > changing buddywatch state on so many buddies so quickly. Also, the > cpu usage is pegged at 100% for around 3 minutes after it reboots, > FWIW. > > Anyone else experiencing rebbots on the 601? Advice is really needed! > > Thanks > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Redundant * servers
Err, what happens if someone transfers a call and the new call leg gets routed through a different asterisk server because the dns changed? Andrew Latham wrote: Use round robin on DNS with a replicated DB on each server On 4/16/07, J. Oquendo <[EMAIL PROTECTED]> wrote: Without using Dundi or SER, any thoughts on the following anyone? Has something similar been implemented anywhere so as to me not having to horribly butcher code... 4 servers SIP1-4 User1 -- -- SIP1 -- \ /\ User2 -- Go to register --- SIP2 - Whereis? --> DB / \/ User3 -- -- SIP3 -- Where users no matter who they are, register and are passed off to the next server in sequence... For example, ten people are all registering right now... User1 --> SIP1 User2 --> SIP2 User3 --> SIP3 User4 --> SIP1 And so on... where an ATA, VoIP phone, etc., would have its information stored via database and pulled and pushed anytime something happened with that User... Make sense? Think of a "load balanced" SIP cluster if you will WITHOUT SER or Dundi... -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-Java website
Well, it _was_ up again Friday, and now it's down again Monday! :( Moises Silva wrote: Hum, I know Stefan, he is an asterisk-java dev, but he is not online right now, I will let him know ASAP. Thanks! On 4/12/07, Doug Garstang <[EMAIL PROTECTED]> wrote: Does anyone know who maintains the Asterisk-java web site at asterisk-java.org? The site seems to have been unavailable for a couple of days now. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-Java website
Does anyone know who maintains the Asterisk-java web site at asterisk-java.org? The site seems to have been unavailable for a couple of days now. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan Streaming
Steve, I was hoping for something native to Asterisk, ie something not requiring a new process. Steve Totaro wrote: Madplay Doug Garstang wrote: Oh poo. No one seems to know. :( Doug Garstang wrote: All, Is there a dial plan command that can stream uncompressed audio from another source? I see there's an MP3Player command that can stream, but I assume that plays MP3's, which means it has to decode them. I'm looking for something that could play .wav or .ulaw (g711) streams. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan Streaming
Eric "ManxPower" Wieling wrote: Doug Garstang wrote: Oh poo. No one seems to know. :( Doug Garstang wrote: All, Is there a dial plan command that can stream uncompressed audio from another source? I see there's an MP3Player command that can stream, but I assume that plays MP3's, which means it has to decode them. I'm looking for something that could play .wav or .ulaw (g711) streams. It would depend on how you define "streaming". If you mean take an audio stream from some web site, the answer is "no, but you might be able to hack some up". If you mean "stream a wav file to a user then the answer is "show application playback" in the Asterisk CLI. Playback supports any file format Asterisk supports. ___ I define streaming as reading an audio stream across the network, not from local disk, such as with the Playback, Read and Background commands. It can't be done? What about the ices2 app? I guess that just plays mp3 and ogg vorbis files. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan Streaming
Stephen Bosch wrote: Doug Garstang wrote: Oh poo. No one seems to know. :( Your mistake is replying to an existing thread and changing the subject line instead of starting a new one. Start a new thread, and people are more likely not only to notice your message, but reply to it. If I had started a new thread, with a slightly modified subject, and asked the question again, right now I'd be justifying my life against the lynch mob for doing so. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan Streaming
Oh poo. No one seems to know. :( Doug Garstang wrote: All, Is there a dial plan command that can stream uncompressed audio from another source? I see there's an MP3Player command that can stream, but I assume that plays MP3's, which means it has to decode them. I'm looking for something that could play .wav or .ulaw (g711) streams. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Dialplan Streaming
All, Is there a dial plan command that can stream uncompressed audio from another source? I see there's an MP3Player command that can stream, but I assume that plays MP3's, which means it has to decode them. I'm looking for something that could play .wav or .ulaw (g711) streams. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Coaching in asterisk
We used ChanSpy to allow a supervisor to listen in on the calls of their staff. There was one huge problem with this, which I imagine would affect whisper as well. The supervisor typically sat fairly close to the worker, and could hear both the voice of the worker as they spoke AND the delayed voice coming through their head phones. It was rather distracting and made it difficult to really be practical. Doug. Dean Collins wrote: Yep, it's called Whisper Check in voip-info.org I think I've read stuff about it there. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Thursday, 8 March 2007 4:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Coaching in asterisk Is there a way to setup a conference where party A can coach another Party B, at the same time, all other parties cannot hear party A? In order words, partis A and B can hear every one, and party A can only be heard by party B. Thnx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Read() status?
Yuan LIU wrote: Does application Read() return a status? Console displays stuff, but show application read doesn't mention any status variable. Yuan Liu I know that read() on a non-existent sound file will cause dial plan execution to abruptly stop (unlike background())... which is very bad imho. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Java w/ Threads
Jesus Mogollon wrote: The best option would be to use AstManProxy and connect your event manager to it. I tried this. Two problems... The Asterisk Manager Proxy sends out a banner of 'Asterisk Manager Proxy/1.2' whereas the Asterisk-Java interface expects to see 'Asterisk Call Manager 1.0' (or something similar, the point is that the banner is different). So, I changed what asterisk proxy manager sends in the source. This allowed Asterisk-Java to connect... and then the Proxy manager went and core dumped . It won't work anyway. The proxy manager prefixes the name of the system to each line of output. The Asterisk-Java interface is not expecting this and will barf. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Java w/ Threads
Stefan Reuter wrote: Jesus Mogollon wrote: The best option would be to use AstManProxy and connect your event manager to it. why would adding a new system in between be better than directly connecting to multiple Asterisk servers? =Stefan Simple. With the manager proxy in between, it does all the hard work of managing all the connections. It's what it's good at, and if it's doing it, I don't have to re-write all the thread management stuff again. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Java w/ Threads
Ok, so I ain't much of a Java programmer, but... Can the Asterisk Java API be written with threads? Ie, I need to connect to multiple Asterisk systems from the one java application. I tried to make my class which implements ManagerEventListener, also implement Runnable, but got errors because the Runnable interface doesn't throw InterruptedException. Anywho... Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Java and Astmanproxy
Has anyone used talked to astmanproxy with the Asterisk Java Manager interface? First suspiscions are that it will not work. Astmanproxy sends a connection banner of 'Asterisk Call Manager Proxy/1.21' which is not what Asterisk Java is expecting. Also, astmanproxy preprends the name of the host to the front of each line of output, and that will break Asterisk Java... I think... anyone done this? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No sound with Playback() or Background()
I goofed that up on my dCAP exam. Spent 20 valuable minutes trying to
fix it!
Eric "ManxPower" Wieling wrote:
This can happen if you have a Digium card (maybe Sangoma too) in the
system that is configured, but has no actual line plugged into it. I
don't know if this applies to analog, but I know it applies to
T-1/PRI/E-1
Kuba wrote:
Joanna Liza Mariazeta wrote:
Perhaps you can add
NoOp(${PLAYBACKSTATUS}) after each Playback, it should return either
"FAILED" or "SUCCESS" in the CLI
Hi Joanna,
I added that, but it looks like it does nothing :(. I don't see any
status after Playback in the CLI.
All I get is:
-- Executing Answer("SIP/206-081af4c8", "") in new stack
-- Executing Playback("SIP/206-081af4c8", "demo-echotest") in new stack
-- Playing 'demo-echotest' (language 'en')
Then, when I hang up
== Spawn extension (12ga, 600, 2) exited non-zero on 'SIP/206-081af4c8'
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Macros, Background(), Return Values...
I am programming a very large dialplan right now (Asterisk 1.4), and a couple of things are annoying the heck out of me. 1. When in a macro, background() does not work properly. If you use the background() app inside a macro, and then press a key, execution returns back to the calling context where it tries to match that extension. I believe this is a known bug. 2. Is there any way to have macros return a value? I can pass arguments to macro's with ARG1..ARGN, but the only way to set a return variable is to set a channel variable. Essentially, I have a large number of global variables which is never good. 3. If you use Gosub to and Return to jump into and out of contexts, and use them like macros to get around the background() problem, the global variable issue becomes worse as there's no way to explicitly pass variables to the contexts when you do this. 4.Every time you make a decision, you have to use GotoIf, which means more code to do simple things like set variables, or do things based on a decision. It would be great if there was SetIf(), MacroIf(), or even doIf() applications. Has anyone tried to program large complex dialplans before and come across some of these issues? How did you resolve them? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queues Problem
You need operator=yes as well... John Breen wrote: Help! I'm (still) having issues with Asterisk Queues. I want to implement a queue so that callers get the 'all our staff are busy at the moment, your call has been placed in a queue and will be answered by the first available operator. You may press 1 at any time to leave a voicemail' announcement, then they can press 1 and leave a voicemail. Documentation with * 1.2.x (I'm using 1.2.15) and in the O'Reilly Asterisk book says I can add a line context=blah to the queue definition and this becomes the 'escape context' where pressing buttons will take you to whilst in the queue. I've done this, and put the relevant context in extensions.conf and put extension 1 in there - and nothing happens - I call into the queue and press 1 and don't go anywhere. Please help if you know how to solve this issue, I have been working on it for a week and it's becoming quite urgent (not to mention causing me to tear my hair out with frustration...) Regards, John Breen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] REGEX Function
What's wrong with this?
exten => s,n,Set(CIDNUM=16505551212)
exten => s,n,Set(foo=${REGEX("^[0-9]+$" CIDNUM)})
This always returns 0, false. That isn't correct.
I also tried:
exten => s,n,Set(dollar=$)
exten => s,n,Set(foo=${REGEX("^[0-9]+${dollar}" CIDNUM)})
and that didn't work either. I am trying to use a regex to see if the
caller id contains numbers only.
Doug.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SayUnixTime Alternate Path?
Does anyone know how I could get the SayUnixTime application to say files from a different sound directory? It looks like it uses the language as a base to determine where to play sound files from. I need to override that. Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
