Re: [Asterisk-Users] Outgoing calls via Sipgate
On Monday 13 March 2006 20:47, Dave Hope wrote:
Hello all,
With some help from people in #asterisk on freenode, I've managed to get
incoming SIP calls working.
Outgoing calls however are however a different matter. My whole working
(incoming calls only) SIPgate configuration can be found here. [1]
When I uncommon what's in there, nothing works. There doesn't appear to
be any useful error being logged , even when debug is enabled for
console and file logs.
If anyone could take a look and show me what needs adding in order for
outgoing calls to work, that would be superb!
My long term goal is to get asterisk running at home, and then persuade
the boss to ditch the Avaya setup we have at the office. But since I'd
likely be the one implementing it, I want to try and get something
working before I commit myself :)
Thanks!,
Dave.
[1] http://files.davehope.co.uk/home.tar
Hi!
I think it was a bad idea to make people download a tar just to help you...
anyway, I did it and my first piece of advice would be, that you should
implement something where the user should dial a 0 for an outside line. So in
the dialplan you would have something like this:
exten = _0X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60)
exten = _0X.,2,Hangup
So now you have to dial a 0, then the number you want to call and so it goes
out over the sipgate account... What's different here is that it's _0X., (see
the dot)? That should make a little difference.
Hope it helps!
Christoph
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Re: [Asterisk-Users] Cannot boot machine up after working on zaptel....
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 28.02.2006, at 15:44, Chris Earle ((CBL)) wrote: Hi all, hard for me to explain this, but it keeps happening on a number of machines I attempt to upgrade zaptel, or do something to zaptel modules. and then I reboot the machine, and for whatever reason, it hangs on loading the modules Either the install wasn't complete, the zaptel modules settings are wrong, whatever but the problem is now I can't get past the boot up and the machine is basically lost Is there any way to bypass the module load attempt or anything? I've tried holding SHIFT down to get the LILO menu, and loading LinuxOLD, but no go I'm on Debian 2.4.18, with Zaptel 1.0.9.2 I understand that there was something wrong in the modules config, but surely I should be able to bypass and get back in to fix it! Any ideas greatly appreciated, as I would rather not have to use an old clone drive and start over Hi Chris, How about you use a Live CD distribution and disable the loading of the driver in some config? Unfortunately I'm not very familiar with Debian, in Gentoo you would edit /etc/modules.autoload/kernel-2.6 and then uncomment the line that loads the module. You should then be able to boot normally and do what you have to do in order to get it to work. Does this also happen when you load the driver using modprobe? Christoph - -- GPG Key ID: 33D6AA8C AIM: zeitgeist2600 ICQ: 271512600 Jabber: [EMAIL PROTECTED] http://www.geisterstunde.org http://www.ceicke.de -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFEBGzw8e/ZGTPWqowRAleSAJ0WIcjiORoRTnd1mTWJNYUj9WuWDACfX7zn 8cadgA0CfHPAgB0Rww5XCHw= =AT6i -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.2 in production w/100+ phones?
On Wednesday 18 January 2006 18:22, Peder @ NetworkOblivion wrote: Is anybody using 1.2 (or 1.2.1) in a production network using Realtime (voicemail, sip or extensions) with 100+ SIP phones? If so, what are your experiences? We've been running 1.0.3 for about a year and it's been rock-solid. We'd like to upgrade to Realtime and 1.2, but I'm afraid of killing our stability. Obviously, we'd do it in stages (upgrade to 1.2, then realtime voicemail, etc), but I'm not sure if 1.2.1 is ready for primetime yet. Thanks. ... never touch a running system... I wouldn't upgrade if there wasn't any great new features that you cannot live without... Peder -- GPG Key ID: 33D6AA8C AIM: zeitgeist2600 ICQ: 271512600 Jabber: [EMAIL PROTECTED] http://www.geisterstunde.org http://www.ceicke.de pgp8e0n7t5YAX.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] digits won't play
Hi! I have a strange problem. In an AGI I tell Asterisk to playback a number, for example 31. I then use the AGI SAY NUMBER command and I only hear thirty and then get: -- Playing 'digits/30' (language 'de') Sep 29 11:47:40 WARNING[3401]: file.c:475 ast_openstream: File does not exist in any format Sep 29 11:47:40 WARNING[3401]: file.c:787 ast_streamfile: Unable to open (format alaw): No such file or directory Sep 29 11:47:40 WARNING[3401]: file.c:475 ast_openstream: File digits/1N does not exist in any format Sep 29 11:47:40 WARNING[3401]: file.c:787 ast_streamfile: Unable to open digits/1N (format alaw): No such file or directory Sep 29 11:47:40 WARNING[3401]: file.c:475 ast_openstream: File does not exist in any format Sep 29 11:47:40 WARNING[3401]: file.c:787 ast_streamfile: Unable to open (format alaw): No such file or directory I have looked inside of /var/lib/asterisk/sounds/digits and all files are present... does it have to do anything with the language 'de'? Where do I change that? Thanks, Christoph ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] digits won't play
It was indeed the problem with the language 'de' setting, setting the SIP client to US gives me the numbers. On Thursday 29 September 2005 12:00, Christoph Eicke wrote: Hi! I have a strange problem. In an AGI I tell Asterisk to playback a number, for example 31. I then use the AGI SAY NUMBER command and I only hear thirty and then get: -- Playing 'digits/30' (language 'de') Sep 29 11:47:40 WARNING[3401]: file.c:475 ast_openstream: File does not exist in any format Sep 29 11:47:40 WARNING[3401]: file.c:787 ast_streamfile: Unable to open (format alaw): No such file or directory Sep 29 11:47:40 WARNING[3401]: file.c:475 ast_openstream: File digits/1N does not exist in any format Sep 29 11:47:40 WARNING[3401]: file.c:787 ast_streamfile: Unable to open digits/1N (format alaw): No such file or directory Sep 29 11:47:40 WARNING[3401]: file.c:475 ast_openstream: File does not exist in any format Sep 29 11:47:40 WARNING[3401]: file.c:787 ast_streamfile: Unable to open (format alaw): No such file or directory I have looked inside of /var/lib/asterisk/sounds/digits and all files are present... does it have to do anything with the language 'de'? Where do I change that? Thanks, Christoph ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] analogue phone with asterisk
An interesting read if you still have an analogue phone that does not speak DTMF tones (yet): http://www.geisterstunde.org/drupal/?q=w48_asterisk On Tuesday 27 September 2005 19:10, Rajesh Bhairampally wrote: I am a newbee to asterisk. I recently installed [EMAIL PROTECTED] Everything went well and my set up is running fine with soft phones, such as kphone and XtenLite. Now, i want to be able to connect my analogue phones to my asterisk pbx box and use it as if i make a regular Phone call (I do have my PSTN gateway account with broadvoice.com and already configured to route through it). I do NOT have a PSTN phone connection. I want to use my analogue phones as the end points for my asterisk box to make and receive calls. All i want is to use my analogue phones instead of soft phones. Can some one help me what hardware interface i need for that and how should i go about it? if there is any HOW-TO for that it will be of great help. thanks, rajesh ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk on windows
On Wednesday 28 September 2005 14:14, Kanishka Somaratne wrote: why can't we compile the asterisk coading in windows, it's done in c++ so it's written in C... have you bothered to look at the source code? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk on windows
On Wednesday 28 September 2005 14:14, Kanishka Somaratne wrote: why can't we compile the asterisk coading in windows, it's done in c++ so it should work in windows as well oh, and did you try google? how about this: http://www.digium.com/index.php?menu=astwind it's a bit of a cheat though 'cause its using coLinux ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Which codec?
On Friday 23 September 2005 11:19, Dan Journo wrote: Is there a guy somewhere on how much bandwidth each codec uses, along with the advantages and disadvantages of each one? Dan Journo calculate it yourself: http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] IAX2 registration
Hi! I have the following setup: PSTN1 Asterisk1 --- IAX2 --- Asterisk2 PSTN2 As you can see, two Asterisk machines are connected via IAX2. There are users connected to each Asterisk machine over a local LAN. Each of these users in both LANs should be able to either use PSTN1 or PSTN2, depending on which extension they dial, no problem here. My question now is, does each Asterisk server need to register with the other Asterisk machine, or is it sufficient if only one Asterisk registers with the other one? Will the other one then know about the connection and be able to make phone calls over that one? Thanks, Christoph ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk don't start
it looks to me like you haven't loaded the zaptel.o or ztdummy.o kernel modules... On Thursday 15 September 2005 16:10, [EMAIL PROTECTED] wrote: Asterisk don't running, because show this message WARNING[6949]: chan_sip.c:8865 reload_config: Section 'authentication' lacks type WARNING[6949]: chan_iax2.c:7491 load_module: Unable to open IAX timing interface: No such file or directory WARNING[6949]: chan_skinny.c:2587 reload_config: Unable to get our IP address, Skinny disabled WARNING[6949]: chan_oss.c:239 sound_thread: Read error on sound device: Resource temporarily unavailable WARNING[6949]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: pri_suspend_acknowledge Sep 15 16:05:46 WARNING[6949]: loader.c:440 load_modules: Loading module chan_zap.so failed! Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] call restrictions
you really should read about the concept of a context in extension.conf, that will answer your question and is also a basic key to understanding Asterisk. http://www.voip-info.org is your friend. Christoph On Wednesday 14 September 2005 10:47, Erdem HAKİ wrote: Hello, I want to use call restriction option. For example, there are 3 registered numbers that 100,200 and 300.I want 100 to call 200 but not 300, btw 300 can call both 100 and 200. How can i configure this? Thanks. Erdem HAKI ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] (no subject)
unless you show us some config files, I doubt that anybody can help you... On Wednesday 14 September 2005 16:46, Pablo Allietti wrote: hi all, i have a box with a te110p and a pbx siemens... connect both with a e1. with a xten soft i can call extensions numbers in my office example 100 102 etc. but when i truy to go outside with the 9 before the call rings in the first extensions (100). this is a asterisk problem? or a pbx problem? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 2 box single Asterisk
Here's my suggestion. Do a dialplan thing where when all trunks on boxA are busy, they are sent via IAX to boxB which sends them out via the ISDN trunks... this way boxA will be your primary box and boxB is your spare box that takes over if everything else is busy... On Tuesday 13 September 2005 10:00, Asterisk Sales wrote: hello list, i need to setup an asterisk system with 5 ISDN trunks. i found C4 cards but they are very expensive. i found that if i use 5 AVM Fritz! cards it would be very cheap. i want to use 2 boxes. 3 in boxA +2 in boxB =5 isdn. and i want, this two boxs to work as a single box so that one box can share ISDN hardware from other box. this system will be serving a call center. currenly we are using a panasonic PBX system but it is driving us crazy. we want to keep the existing pbx setup and add asterisk with it to handle the call center operations. we also need to communicate with pbx users from Asterisk. our pbx has 6 analog trunks. so we can use TDM400P please help how can i solve this situation will low cost and performance. best regards shaon ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] PHP and ASterisk Manager
I looked into the source code of Asterisk to figure out how the printf()
statements were spaced. That's the power of open source, you can look under
the hood for these questions. It's easy to find, even for non-C-Gurus. Just
do a grep for the string that you want inside of the Asterisk source
directory and it will give you the file that the string you are looking for
is in. Then simply open the file, search for the string and look at the
printf() statement.
Christoph
On Tuesday 06 September 2005 21:16, Anton Krall wrote:
I was able to do and if and while loops to get the block of lines I want..
Now.. Another issue.
I need to parse the line read to insert it into a table but seems Asterisk
inserts TABS or SPACES inconsistantly.. For example:
Xxx(TAB)xxx(5 spaces)xxx
Next line
Xxx(TAB)xxx(3 spaces)xxx
Im having a hard time figuring out how Asterisk Manager returns the stuff
:)
Well..s o far so good...
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Matthew Boehm
|Sent: Martes, 06 de Septiembre de 2005 01:49 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] PHP and ASterisk Manager
|
|Anton Krall wrote:
| Guys, is anybody using PHP sockets to connect to the Manager
|
|and send
|
| command like show voicemail users for example or any other?
|
| My question is, how to parse the return info in a way that can be
| shown back to the user via web (discard all the manager
|
|responses not needed)?
|
|Use preg_match() to match the lines you want the user to see
|on the website.
|
|$socket = fsockopen(localhost,5038, $errno, $errstr, 30);
|
|if(!$socket) {
| print No socket;
| exit();
|}
|
|fputs($socket, Action: Login\r\n);
|fputs($socket, Events: Off\r\n);
|fputs($socket, UserName: bleh\r\n);
|fputs($socket, Secret: bleh\r\n\r\n);
|
|fputs($socket, Action: Command\r\n);
|fputs($socket, Command: show channels\r\n\r\n);
|
|fputs($socket, Action: Logoff\r\n\r\n);
|
|while(!feof($socket)) {
| $buff = fgets($socket,1024);
| if(preg_match(/SIP\/.*/, $buff)) {
| print I found a SIP call;
| }
|}
|
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Re: [Asterisk-Users] Ethernet / TcpIp phones
try google for VoIP Phone ;-) or here: http://www.voip-info.org/tiki-index.php?page=Asterisk+phones On Wednesday 07 September 2005 11:19, Alex wrote: Is there any VoIP phones available which can be plugged directly to the Ethernet network? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] unresolved symbol when loading ztdummy
On Tuesday 30 August 2005 17:01, Braz wrote: Your kernel has to be compile with CONFIG_CRC_CCITT=y or m. I couldn't find that option in the kernel, but inserting the zaptel module before ztdummy works of course. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] unresolved symbol when loading ztdummy
On Wednesday 31 August 2005 11:11, Sergio Serrano wrote: This option is under Library routines in your kernel configuration. ah, yes. In kernel 2.6.* it is. not in 2.4.26 ;-) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] strange problem
Hi! I have a strange problem with Asterisk (Asterisk 1.0.8-BRIstuffed-0.2.0-RC7k) on a VIA Samuel x86. When I make a call from the CLI either over IAX2 or SIP, my first call after the initial start of Asterisk works fine, even though upons starting Asterisk tells me Read error on sound device: Resource temporarily unavailable. I hear the call over the loudspeaker. When I hang up (using CLI hangup command) and try to place another call, I get a busy signal over the loudspeaker and I get Error reading from sound device (If you're running 'artsd' then kill it): Resource temporarily unavailable. If I issue a hangup again, I can dial out fine again. I don't have artsd running and nothing else besides Asterisk is using /dev/dsp (according to lsof). I have read somewhere that this might have to do with the sound card chip that I'm running (VIA Technologies, Inc. VT82C686 AC97 Audio Controller (rev 50)). Unfortunately I don't have the luxury of getting to see the debug output of /var/log/asterisk/debug on that machine, because it runs on a 128MB read only file system. I'm not loading chan_alsa.so, only chan_oss.so as I think this might have something to do with the problem. Any help would be great, or any hints into a possible direction. Thanks, Christoph ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] unresolved symbol when loading ztdummy
Hi! When I try to load the ztdummy driver via insmod ztdummy, I get the following errors: /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_unregister /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_transmit /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_receive /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_register I'm using zaptel-1.0.8 and a 2.4.26 kernel. Unfortunately I cannot change to a more recent kernel, and another drawback that I have is, that I cannot compile on the machine that Asterisk is supposed to run on. However I booted my development machine with a 2.4.26 kernel and the compilation went well. What I am actually trying to archive is, getting the ztdummy driver to work in order for my Asterisk machine to do IAX2 calls. As far as I have understood it this is necessary if I don't have any Digium hardware. What is the problem that I'm having and how can I fix it? Thanks a lot, Christoph ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] 1-800 number
On Thursday 18 August 2005 22:27, Matt Hess wrote: Just call a milliwatt..? you have a number? I'm also willing to pay my regular fees to my provider for those 3-4 minutes of testing. ___ 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] 1-800 number
I'm trying to solve my Nikotel problem (see previous post) where the problem is that I get a hangup after 2 minutes, therefore I need some number that doesn't cost anything and gives me some audio for a long time... On Thursday 18 August 2005 15:08, Jonathan k. Creasy wrote: What problem are you trying to solve with this? Just stepping out on a limb but it sounds like you are trying to swat a fly with an F-16. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christoph Eicke Sent: Wednesday, August 17, 2005 4:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 1-800 number Hi! I'm searching for a 1-800 number that simply plays music for a long time (3mins) and no one picks up. I've bothered the ATT lines so far when trying out my SIP-PSTN connection but then always someone answered :-) Anyone have a number? Christoph ___ 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 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] Nikotel issues
Hi!
I've read in the archives that there are problems concerning Nikotel calls
being disconnected after two minutes. I had the same problem yesterday. Is
there a fix? There was only a giving up statement after the last e-mail in
the archive, I'm about to do that too.
Here's my sip.conf entry for Nikotel (left out the register stuff 'cause it's
working):
[nikotel]
type=friend
host=calamar0.nikotel.com
username=user
secret=pass
fromuser=user
fromdomain=nikotel.com
qualify=yes
context=nikotel-incoming
insecure=very
canreinvite=no
promiscredir=yes
diallow=all
allow=alaw
allow=ulaw
allow=gsm
extension.conf:
[nikotel-incoming]
exten = 3740525,1,NoOp(Invoming call via nikotel-us)
exten = 3740525,2,Dial(IAX2/christophSIP/30${CONSOLE},30)
exten = 3740525,3,VoiceMail(u30)
exten = 3740525,4,Hangup
Thanks for any help,
Christoph
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[Asterisk-Users] 1-800 number
Hi! I'm searching for a 1-800 number that simply plays music for a long time (3mins) and no one picks up. I've bothered the ATT lines so far when trying out my SIP-PSTN connection but then always someone answered :-) Anyone have a number? Christoph ___ 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] Voicemail Retrival
On Wednesday 17 August 2005 10:29, Sharadindu Mohanty wrote:
Hi,
Hi!
Any ideas??
Yes, I do it in the following way. In extension.conf add this line:
exten = ,1,VoiceMailMain(s${CALLERIDNUM})
exten = ,2,Hangup()
Here any extension can call and then automatically gets directed to their
voicemail where they have some options.
I hope this helps,
Christoph
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Re: [Asterisk-Users] 1-800 number
On Wednesday 17 August 2005 10:45, Michael K. Rodriguez wrote: More info I don't quiet understand your mail ;-) Do you want more info from me? ___ 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] problem with sound device
On Monday 15 August 2005 21:08, Innocent Evil wrote: I am getting this whenever I start asterisk. Aug 15 15:03:58 WARNING[11754] chan_oss.c: Read error on sound device: Resource temporarily unavailable sounds like your soundbard is blocked by another program. Sometimes applications like KDE or XMMS block the sound card, even after these are turned off. It then takes a while for the soundbard to become available again. Christoph ___ 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] permission denied when monitoring channel OSS/dsp
Hi! When I want to monitor the OSS/dsp channel through the Asterisk management interface, I get a permission denied error: Action: Monitor Monitor: OSS/dsp File: 1124096949 Mix: 1 Response: Error Message: Permission denied My permissions for /var/spool/asterisk look like this: pound:~# ls -la /var/spool/asterisk/ total 40 drwxr-xr-x 10 asterisk asterisk 4096 Aug 9 10:19 . drwxr-xr-x 9 root root 4096 Aug 4 14:40 .. drwxr-xr-x 2 root root 4096 Jul 8 09:40 dictate drwxr-xr-x 2 root root 4096 Jul 8 09:40 meetme drwxr-x--- 2 asterisk asterisk 4096 Aug 9 10:32 monitor drwx-- 2 root root 4096 Jul 4 16:56 outgoing drwxr- 2 root root 4096 Jul 4 16:56 qcall drwxr-xr-x 2 root root 4096 Jul 8 09:40 system drwxr-xr-x 2 asterisk asterisk 4096 Mar 21 12:23 tmp lrwxrwxrwx 1 root root 37 Jul 13 10:08 vm - /var/spool/asterisk/voicemail/default drwxr-xr-x 3 asterisk asterisk 4096 Jul 4 16:53 voicemail So, there shouldn't be any problems writing to disk. Anything else that I need to take into account, especially something special about monitoring the OSS/dsp channel? Thanks, Christoph ___ 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] permission denied when monitoring channel OSS/dsp
On Monday 15 August 2005 11:11, Christoph Eicke wrote: Hi! When I want to monitor the OSS/dsp channel through the Asterisk management interface, I get a permission denied error: Action: Monitor Monitor: OSS/dsp File: 1124096949 Mix: 1 Response: Error Message: Permission denied it's always nice to answer your own posts ;-) The Permission denied message had nothing to do with file attributes, but with what is written inside of the /etc/asterisk/manager.conf users's section that is connecting to the Asterisk management interface. There you have to set the right permissions. ___ 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] txgain for SIP?
Is there an option txgain for SIP in Asterisk? My users all complain that their other parties think that they are way too silent even though they all have their mic volume all the way up and also enabled the 'mic boost' option. This happens with all the clients that we're using and also with different model headsets, so my last hope is txgain for SIP. Thanks Christoph ___ 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] Billion BRI PCI card
On Friday 12 August 2005 09:43, John Fawcett wrote: when using in NT mode does the card require additional power or is it able to supply enough power by itself to the S0 bus? I don't know the exact specifics about the Billion card, but I have a setup where I have an extra NTBA connected to the ISDN card (S0 bus coming from ISDN card into NTBA, cable needs to be crossed) and then my ISDN phones are connected to the S0 bus coming out of the NTBA. I assume that the Billion card works in a similar fashion and doesn't have the necessary power to directly connect phones to it. I hope that this helps you a little. ___ 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] OT: Sendmail question
On Friday 12 August 2005 11:46, Rudolf Ladyzhenskii wrote: Old messages are in the queue. I can see sendmail is trying to talk to the remote mail server, but never gets a responce and times out. So message stays in the queue. You should try and deliver them yourself. It sounds to me like sendmail is configured to relay messages using a smart host (the one it's trying to talk to). Do you maybe need to log in to that host in order to send mail? You can try this out yourself: set the DNS name of the mail server it relays it to in /etc/hosts to your local host, don't start sendmail as a server daemon (shouldn't bind to port 25) and do a nc -lt -p 25 and look what it's trying to do when sending mail via the command line. Christoph ___ 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] CLI and Dial
On Wednesday 10 August 2005 16:43, Esben Stien wrote: Moises Silva [EMAIL PROTECTED] writes: make sure you have the next line in /etc/asterisk/modules.conf load = app_dial.so Not only that, but be sure to have a sound system loaded in the modules.conf files. there were actually multiple problems, firstly chan_alsa.so wasn't loaded, secondly the Gentoo init script added some options to the Asterisk startup that made the 'dial' command disappear. In /etc/init.d/asterisk you just needed to comment those option things out and then it worked fine. ___ 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] CLI and Dial
Hi! I have two Asterisk installations, one being a 1.09 bristuff installation and one 1.08 installation. In the 1.09 installation I have the Dial command available on the CLI, in the 1.08 installation I don't. My question is now: was that a new feature in 1.09 or is it a bristuff specific thing or is my 1.08 installation simply lacking features? Thanks, Christoph ___ 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: Abwesenheitsnotiz: [Asterisk-Users] Nortel Option 11 and TE110P o f Digium
On Friday 05 August 2005 19:29, Alvaro Parres wrote: ??? i dont understand. it's an out-of-office reply until august 9th On 8/5/05, Siegel, Joerg [EMAIL PROTECTED] wrote: Ich bin am 9.8. wieder im Hause! Mit freundlichen Grüßen, Jörg Siegel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is there a right place for a include_once statement in a PHP AGI script?
On Friday 05 August 2005 14:04, Leo Burd wrote: Hello there, I'm new to PHP AGIs and I'm having problems with a particular script that has a include_once statement on it. If I remove that stament, the script runs until the section of the code that depends on the include and then returns. If I include that statement, the script does not seem to run at all. What shall I do? Leo, wrap a function around whatever is in the included script, make your include_once() statement at the top of the AGI and then simply call the function at the place where it's necessary for that code to be executed. Christoph ___ 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] USB ISDN devices
Has anyone had any luck getting USB ISDN devices to work with Asterisk? I have bought a DayTrek miniVigor 128 and would like to get it to work with CAPI or mISDN. Has anyone every successfully done something like this? Thanks, Christoph ___ 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] ABI manager - redirect
I'm very interested in the redirect feature of Asterisk. So far I haven't gotten it to work. My scenario is that there is a two party call going on where I want to send one of those parties somewhere else. In the wiki is only an example how to send both parties to a meetme room. Is the ExtraChannel parameter required? This is what I have: Action: Redirect Channel: SIP/8080-e2a7 Exten: 5000 Context: local Priority: 1 Now, the SIP/8080 channel is connected to one extension, if I now redirect it to extension 5000 (as shown above), the SIP channel hears a short ring tone and then connects back to it's original extension which starts over again. Both, the extension that the SIP/8080 channel is connected to and where it should be redirected to are in the same context (local). Any hints would be great, Thanks, Christoph ___ 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] VoiceMailMain issue..
On Monday 25 July 2005 09:48, Mauro Zanin wrote:
Hi everybody,
Hi Mauro!
I'm in a middle of a Asterisk learning period. I am at a very good point
except I'm not able to use VoiceMailMain.
This Is my simple dialplan regarding VoiceMail
;Number that the IP Phones dial to access voice mail
exten = 22999,1,VoiceMailMain (s${CALLERIDNUM})
exten = 22999,2,Wait(3)
exten = 22999,3,Hangup
your dialplan looks good
Why do I get Forbidden 403 and one console display :
Jul 25 09:48:09 WARNING[1117207472]: pbx.c:1274 pbx_extension_helper: No
application 'VoiceMailMain ' for extension (home, 22999, 1)
Anybody knows why?
Have you checked /usr/lib/asterisk/modules/ and made sure that
app_voicemail.so is there?
Christoph
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Re: [Asterisk-Users] Asterisk Configuration
On Monday 25 July 2005 14:07, Afzaal Mirza wrote: I am new to this mailing list. Can someone send me a guide or steps to configure Asterisk on Linux box? I will highly appreciate. http://www.voip-info.org ___ 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] a ne pas voir
On Thursday 21 July 2005 12:41, ali kia wrote: hi all i suggest to create a goup in hotmail in order to discuss any problem on line in msn i think it's more practical than e-mail group also I would prefer not to switch to something M$ based... ___ 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] Problems installing asterisk-addons
On Thursday 21 July 2005 15:28, Angus Comber wrote: My asterisk version is Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j It is a version put together by Junghanns.net - for working with their ISDN cards. Mmm I wonder if that is the problem? If so then what version of asterisk-addons do I install. I didn't see anything about asterisk-addons on the junghanns.net site. You are right, that is the problem. I wasn't able to compile the addons with the version from junghanns.net. I suspect that it's because those addons compile the MySQL realtime extension and the Asterisk version coming with the bristuff package has no support for the realtime extension yet. Christoph ___ 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] Mahler's Book - New Project
On Wednesday 20 July 2005 15:56, David Stude wrote: Hi all, I'm currently gearing up for a possible PBX replacement project using Asterisk, and I'm just breaching the iceberg of information that's available. I typically like to have something thick with pages in front of me. Mahler's book was the first one to come up and it seems like a good place to start. However, the big name bookstores tell me it'll take up to three weeks, and this project simply can't endure that wait. Does anyone know where it's possible to get a paper copy *quickly*? I saw earlier today that Paul Mahler's own company sells the book on their website as an ebook... so you could just print it out. BUT... I also found the book very bad. No clear structure, concepts are not really explained and it lacks in depth explanation of functions. It's very shallow. My $0.50 Christoph ___ 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
