Re: [asterisk-users] Upgrade 4 to 8 Analog Lines Question
I would suggest you avoid TDMoE its support is pretty much depreciated and not supported by Digium. Does not work well with Kernel 2.6.x. On 4/9/07, Michelle Dupuis [EMAIL PROTECTED] wrote: Look at a channel bank with MGCP/TDMoE/USB interface (there are plenty around). We too prefer to keep fxs/fxo hardware outside of the * box. Michelle Dupuis Technical Support Specialist Generation Software - Linux and Asterisk solutions and support. Visit us at www.generationd.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Freeze Sent: Monday, April 09, 2007 9:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Upgrade 4 to 8 Analog Lines Question Hello I have an office with a T1 that provides 4 (out of 8) analog PSTN lines thru an adtran board. I want to add 4 more analog lines. Currently I have a Digium TDM40B. I'm wondering what the best upgrade path is, where I define 'best' as the solution that is most likely to work without problems (like interupt conflicts) and work with my current echo tuning . I see my purchase options as follows: 1) TDM40B - use with the current TDM40B 2) Sangoma Remora A20200 - use with the current TDM40B 3) Sangoma Remora A20400 - replace the current TDM40B Any info will be greatly appreciated. Thanks Jim -- Jim Freeze ___ --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] Reverse-ATA : Using PSTN lines to connect toAsterisk
Take a look at these http://www.telephonydepot.com/product_p/105-056-4104.htm http://www.telephonydepot.com/ProductDetails.asp?ProductCode=105%2D056%2D4108 I would suggest you avoid the AudioCodes units, AudioCodes blatantly ignores the GPL and refuses to release even their kernel source. On 4/10/07, Mike [EMAIL PROTECTED] wrote: Thanks Alex, That was my original thought, to just buy a TDM400 from Digium and put in as many FXO as I wanted, but I liked having the ease of just buying something off the shelf, even if it meant paying a little more. But it looks like I won't have much of a choice. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Tuesday, April 10, 2007 10:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Reverse-ATA : Using PSTN lines to connect toAsterisk Hi Mike, You should be looking at ATAs that have FXO, rather than FXS interfaces. Most ATAs come with FXS ports so that you can connect analogue phones to them, but in this case you're wanting to take PSTN lines from the outside, so FXO is desirable. Second, you'd have to make sure that the ATA supports the sort of application you're using it for; most are manufactured on the opposite premise. I am actually not sure offhand of any ATA firmware that I know that I imagine would work this way, although I'm confident it exists as consecutive back-to-back analogue-VoIP adaptations in many scenarios can get quite complex and requires that flexibility. Basically, you're looking for a small IP PBX that uses SIP internally among its private nodes and takes PSTN trunks from the outside. That's what PBXs typically do. :-) If all else fails, you can always roll your own functionality of this nature by using FXO cards in Asterisk. There are various distributions that package it in a very lightweight and reusable manner specifically for this type of purpose, or you can roll your own if it's scalable enough. -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --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] Polycom 330/320
How is the screen compared to the other Polycom products? On 4/9/07, Jessee J Holmes [EMAIL PROTECTED] wrote: Mike, I don't have much information, except they are due for shipment soon (mid to end of April to distribution from Polycom). We've demoed a couple and I personally believe they'll be a tough phone to find in stock for the first few months their released. Demand on these from what I'm seeing right now is very, very high. I think they are a great addition to the family and most importantly they have FULL DUPLEX SPEAKERPHONE! :) 550's are released products though. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Apr 9, 2007, at 3:55 PM, Mike wrote: Ah, thanks. I didn't realize this. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Nickerson Sent: Monday, April 09, 2007 16:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 330/320 Mike [EMAIL PROTECTED] wrote: How do you guys like the 330 and 320? Mike, As far as I am aware, neither of these handsets are presently shipping from Polycom, so most people's experience will be limited to PDF brochures and pretty pictures. On the face of it, this looks like a good alternative to the IP301 since it adds native 802.3af PoE support. Not sure yet exactly where the pricing will slot in, however. -Darren -- Darren Nickerson Senior Sales Support Engineer Telephony Depot www.telephonydepot.com +1.215.825.8710 ext 8106 (office) +1.215.243.8335 (fax) ___ --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
[asterisk-users] Nagios asterisk monitoring
Dear list,
I am trying to configure the nagios plugin called check_sip. I just read the
README file included with the plugin. I follow the readme steps to configure
the plugin, without success. In the nagios web interface I can see (No
output!) In the status information column. If I run the chech_sip plugin
from a linux console, I get
/usr/local/nagios/libexec# ./check_sip -u sip:[EMAIL PROTECTED]
SIP 200 OK: 0.00 second response time
I do not know why If I run the plugin from the consle it works ok, but if I
run it from Nagios web interface it does not run.
Anyone are using this plugin?
Could you helpme to solve that?
Any clue will be appreciated.
Thanks for your time.
VoipCrazy
Here goes my nagios check_sip plugin configuration.
define command{
command_namecheck_sip
command_line$USER1$/check_sip -u $ARG1$ -H $HOSTADDRESS$ -w 5
}
define service{
use generic-service
host_name -PBX
service_description SIP test
check_command check_sip!sip:[EMAIL PROTECTED]
contact_groups admins
max_check_attempts 4
normal_check_interval 5
retry_check_interval1
notification_interval 240
check_period24x7
notification_period 24x7
notification_optionsc,r
}
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[asterisk-users] RE: Zaptel 1.4.1 Install Modules CentOS
Hi List / Tzafrir I can't thank you enough for your support through this problem. I had another look on voip-info.org/wiki at CentOS. There is a good post on installing Astrerisk on CentOS, I was reading it through, and thought I would double check a few things. It turns out the linux symbolic links to the Kernel source were pointing to the wrong version. Somone else who had been on the server before me had tried to install the source but had not correctly identified it was the smp version required. Using some of the knowledge you had shared with me and doged determination it now works. When people post questions asking what distro to use, pick one and stick to it. I'm certain half of my troubles have arisen from using a distro I am not familier with. Althouh Slackware is considered Hard Core by some, it's what I am more used to (and installing from CD my self). Again, many thanks Chris -- Chris Blunt -Original Message- Date: Tue, 10 Apr 2007 19:56:43 +0300 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [asterisk-users] RE: Zaptel 1.4.1 Install Modules CentOS To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Wed, Apr 04, 2007 at 05:52:46PM +0100, Chris Blunt wrote: Hello again I tried the yum install kernel-smp-devel this seemed to download an updated version that was not the same as the version running, so I backed it out using rpm -e kernel-smp-devel I then proceeded to do uname -r to verify the kernel version (output: 2.6.9-42.0.3.ELsmp) and did yum install kernel-smp-devel-2.6.9-42.0.3.EL.i686 If I now do ls -l /lib/modules/`uname -r` I do get build - /usr/src/kernels/2.6.9-42.0.3.EL-smp-i686 I have then tried recompiling zaptel. But same trouble I'm afraid! maybe ztdummy.ko was not regenerated? 'make clean' is normally not needed when changing kernel versions, as Kbuild is usually smart enough to tell the difference. What is the output of: modinfo ./ztdummy.ko -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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: *****SPAM***** [asterisk-users] RE: Zaptel 1.4.1 Install Modules CentOS
Software zur Erkennung von Spam auf dem Rechner priamus.teamware-gmbh.de hat die eingegangene E-mail als mögliche Spam-Nachricht identifiziert. Die ursprüngliche Nachricht wurde an diesen Bericht angehängt, so dass Sie sie anschauen können (falls es doch eine legitime E-Mail ist) oder ähnliche unerwünschte Nachrichten in Zukunft markieren können. Bei Fragen zu diesem Vorgang wenden Sie sich bitte an [EMAIL PROTECTED] Stop it. If you consider it spam, discard it. Don't tell the list. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] IM on x-lite
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all, Currently i'm running asterisk 1.4.2 on fedora core 6 x64, using sip i configured x-lite on client n its runs well. But the instant message won't work. it says a notice the person tou are sending messages to is using an earlier version of x-lite, which does not support rich text and emoticons. you may have to send some of your messages again when i send the message an error occured with message methode not allowed 1. Is it posible to activate instant message service on asterisk using x-lite softphone? how? 2. why my contact's availability always shown offline on x-lite even the user was shown online on asterisk CLI? thanks Sigid -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFGHJ8xqiPNNgPlDu0RArByAKDCQY29M+IpSxYHqUvUKIWZZdW4zQCgi/h4 g/BO8rvY/vA+Wyb/c3gnBI4= =Mlty -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] help with Sipura SPA 3000
On 10 de abr de 2007, at 23:05, James Harper wrote: I've bought a Sipura SPA 3000, and succesfully connected it to my Mac, where I installed Asterisk 1.4.0. Both ports (FXO and FXS) are well configured). However, living in Brazil, I'd like to know if there are optimal settings to my PSTN that I should enter into the config of the device. I experience a little bit of echo on the FXO probably because I raised the gain of that port because I wasn't sounding loud enough. Get the impedance settings right. An impedance mismatch will cause echo (but may not be the only cause) Thanks a lot for your answer!! But, how do I found out what's the correct impedance of lines here? But there are two things I would like to do with the device, and I'd appreciate if anyone could help me out: 1 - Is there a way to stop cutting other people when I speak through the PSTN? What I mean is that, when sound is captured by my telephone, it dimishes the other peer's voice, and sometimes it makes communication harder, as if the line weren't full duplex. I think the 'echo suppression' setting causes this. It is meant to reduce the incoming audio (and hence the echo) while you are talking, which can be annoying but is supposed to be less annoying than the echo itself. I see... 2 - How can I gain full control to the FXS? I mean, a simple * dialed is not sent for asterisk (the server) interpretation, probably because it's used by Sipura's suplementary services, I don't know. Also, is it possible to get a dial tone from ASterisk, instead of Sipura's? My goal with this is to provide users with direct access to the PSTN line pressing 0, instead of collecting calls and making the call themselves, or at least making ignorepat to work! A dialplan of '(S0:s)' will get your phone to jump straight into the 's' extension in asterisk as soon as someone picks it up. From there you can do something like: [sip_ata_incoming] exten = s,1,Answer exten = s,n,DISA(no-password|sip_extension_in) so Asterisk will give you dialtone and do the dialplan stuff for you. From the 'sip_extension_in' context you can make a single '0' or '*' call the PSTN line. I think if I choose the * to get a dialtone it won't work because it seems that the SPA-3000 will pick up that character and use it as if I was trying to access its own services... By the way, for transfering calls, will asterisk or the SPA the one that will actually do the transfer? Good luck with the echo situation. I have an spa3000 and no matter what I do I get echo coming back to me with almost no reduction in volume!!! Thanks... I don't mind if the echo is small, I actually prefer a small echo than that cutting thing... :( Cheers, Francis James ___ --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] how to install asterisk on redhat ?
You should consult with the initial chapters of this book: Asterisk, the future of Telephony (Download from the following link) http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 On 4/11/07, Alex Balashov [EMAIL PROTECTED] wrote: On Wed, 11 Apr 2007, Malik Mulki (Plant, Feed, Makassar) said something to...: Hiasterisk users... how to install asterisk on redhat ? There are numerous installation guides on this subject. But in general, you can either install a contributed RPM, or download the source code and compile it (along with libpri and zapata telephony interface if you need them). Check out ftp://ftp.digium.com/ -- Alex Balashov [EMAIL PROTECTED] ___ --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 -- Regards Rizwan Hisham Software Engineer ___ --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 Pickup with more than one argument
Dear all, I tried to use the following sintax to implement call pickup in Asterisk 1.2.17 with no success: exten = _**5219/5215,1,Pickup(5219) exten = _**5219/5215,2,Pickup(220408108) exten = _**5219/5215,3,Hangup Asterisk seems to just do the first priority command (Pickup(5219)) and if the ringing call comes from the channel 220408108, it doesn't jump to the second priority command. I've also tried to do it in only one line, like this: Pickup(5219220408108) but it doesn't work! After reading in voip-info, only in Asterisk 1.3 development, this issue has been considered to be implemented... I wonder if Asterisk 1.4 implements this since no version 1.3 has been released! Other option seems to be the use of Pickup2, but is it a stable option to implement in a production system? Thanks in advance, Ricardo. ___ --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] calls bridging
dear can asterisk dial two numbers, then bridge them.(like jah jah) best Mani Looking for earth-friendly autos? Browse Top Cars by Green Rating at Yahoo! Autos' Green Center. http://autos.yahoo.com/green_center/ ___ --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] Purposely setting red alarm on PRI for testing purposes
Does anyone know if is possible to purposely set red alarm status on PRI circuit for testing purposes (other than unplugging it). I have looked for a console command which might allow this ___ --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] Re: Play audio and continue to next priority before audio ends...
Alejandro Mejía [EMAIL PROTECTED] wrote: I would like to know how to playback an audio file to the caller, and while it's played asterisk to continue executing the next priorities on extensions.conf That's not the case when using playback command, because the next priority is executed until the audio file ends playing. I want to evaluate some variables while caller hears the audio file. Any ideas? Look at the Background() application. It does just what you are asking for. I'm surprised no-one else has mentioned this. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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: Play audio and continue to next priority before audio ends...
Tony Mountifield wrote: Alejandro Mejía [EMAIL PROTECTED] wrote: I would like to know how to playback an audio file to the caller, and while it's played asterisk to continue executing the next priorities on extensions.conf That's not the case when using playback command, because the next priority is executed until the audio file ends playing. I want to evaluate some variables while caller hears the audio file. Any ideas? Look at the Background() application. It does just what you are asking for. I'm surprised no-one else has mentioned this. I suspect because Background does not execute the following priorities until it has finished playing the file. The original poster said that he/she wanted to continue executing the next priorities while the audio file was paying. Background does not continue to execute priorities while it is playing the file. ___ --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: Play audio and continue to next priority before audio ends...
On Wed, 11 Apr 2007, Tony Mountifield wrote: Alejandro Mejía [EMAIL PROTECTED] wrote: I would like to know how to playback an audio file to the caller, and while it's played asterisk to continue executing the next priorities on extensions.conf That's not the case when using playback command, because the next priority is executed until the audio file ends playing. I want to evaluate some variables while caller hears the audio file. Any ideas? Look at the Background() application. It does just what you are asking for. I'm surprised no-one else has mentioned this. Are you sure it does that? I'm under the impression that it waits until the sound(s) have finished playing before moving on to the next priority. (While listening for digits to be pushed, then be dialled) -= Info about application 'BackGround' =- [Synopsis] Play a file while awaiting extension [Description] Background(filename1[filename2...][|options[|langoverride][|context]]): This application will play the given list of files while waiting for an extension to be dialed by the calling channel. To continue waiting for digits after this application has finished playing files, the WaitExten application should be used. The 'langoverride' option explicity specifies which language to attempt to use for the requested sound files. If a 'context' is specified, this is the dialplan context that this application will use when exiting to a dialed extension. If one of the requested sound files does not exist, call processing will be terminated. 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
RE: [asterisk-users] Re: Play audio and continue to next prioritybefore audio ends...
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Wednesday, April 11, 2007 8:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Play audio and continue to next prioritybefore audio ends... On Wed, 11 Apr 2007, Tony Mountifield wrote: Alejandro Mejía [EMAIL PROTECTED] wrote: I would like to know how to playback an audio file to the caller, and while it's played asterisk to continue executing the next priorities on extensions.conf That's not the case when using playback command, because the next priority is executed until the audio file ends playing. I want to evaluate some variables while caller hears the audio file. Any ideas? Look at the Background() application. It does just what you are asking for. I'm surprised no-one else has mentioned this. Are you sure it does that? I'm under the impression that it waits until the sound(s) have finished playing before moving on to the next priority. (While listening for digits to be pushed, then be dialled) -= Info about application 'BackGround' =- Snip What the OP is requesting is that they are able to perform logic (ie Database Lookups, Parsing of information, etc.) while a sound file is playing. The only way I have been able to do this is with an AGI. Alex ___ --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] Re: nadi: branch 1.4 r61342 - /branches/1.4/channels/chan_misdn.c
Author: nadi Date: Wed Apr 11 05:52:28 2007 New Revision: 61342 URL: http://svn.digium.com/view/asterisk?view=revrev=61342 Log: AOCD's are now exported to asterisk channel variables. Modified: branches/1.4/channels/chan_misdn.c This is very cool, something i've waited for a long time :-). Is there a way to write this into CDR or sendtext() it to a channel? Regards, Andreas _ Live Search delivers results the way you like it. Try live.com now! http://www.live.com ___ --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] how to install asterisk on redhat ?
Malik Mulki (Plant, Feed, Makassar) wrote: Hiasterisk users... how to install asterisk on redhat ? I use the following for CentOS 4.x. Works like a treat: http://www.voip-info.org/wiki/index.php?page=Asterisk+Linux+Centos -- Warm Regards, Lee ___ --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] wideband codec support
Hi, Does Asterisk support wideband VoIP? Is there support for Speex 16 KHz? Thanks, Madhuri Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit. http://farechase.yahoo.com/promo-generic-14795097 ___ --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] wideband codec support
Madhuri Patwardhan wrote: Hi, Does Asterisk support wideband VoIP? Is there support for Speex 16 KHz? What devices support this? As I'm sure you know, if the call hits the PSTN it will always be ulaw or alaw and so will not be wideband. Wideband codecs are only helpful if the call IP end to end. ___ --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: asterisk-users Digest, Vol 33, Issue 33
Looks like he went for a passover cruise somewhere. FB, I hoped you enjoyed it, was it non gebrokts as well? On 4/9/07, Alex Robar [EMAIL PROTECTED] wrote: David, It's not US format. He's away April 4th through April 11th. There was a big discussion about FB and his absence on this list a few days ago. Alex On 4/9/07, David Boyd [EMAIL PROTECTED] wrote: Could someone please remove this person from the list. It seems that the person is saying they will be away for (9) nine months, with their auto-reply set. dave On Mon, 2007-04-09 at 18:26 +0200, [EMAIL PROTECTED] wrote: Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --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] Re: Play audio and continue to next priority before audio ends...
Gordon Henderson wrote: On Wed, 11 Apr 2007, Tony Mountifield wrote: Alejandro Mejía [EMAIL PROTECTED] wrote: I would like to know how to playback an audio file to the caller, and while it's played asterisk to continue executing the next priorities on extensions.conf That's not the case when using playback command, because the next priority is executed until the audio file ends playing. I want to evaluate some variables while caller hears the audio file. Any ideas? Look at the Background() application. It does just what you are asking for. I'm surprised no-one else has mentioned this. Are you sure it does that? I'm under the impression that it waits until the sound(s) have finished playing before moving on to the next priority. (While listening for digits to be pushed, then be dialled) -= Info about application 'BackGround' =- [Synopsis] Play a file while awaiting extension [Description] Background(filename1[filename2...][|options[|langoverride][|context]]): This application will play the given list of files while waiting for an extension to be dialed by the calling channel. To continue waiting for digits after this application has finished playing files, the WaitExten application should be used. The 'langoverride' option explicity specifies which language to attempt to use for the requested sound files. If a 'context' is specified, this is the dialplan context that this application will use when exiting to a dialed extension. If one of the requested sound files does not exist, call processing will be terminated. Gordon You could break up the audio files you want to play into smaller bits and execute dialplan mojo in between the audio chunks. Thanks, Steve ___ --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] wideband codec support
Will a point to point connection between two Polycom HD Voice SoundPoint(r) IP 650 units fit the bill? On 4/11/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Madhuri Patwardhan wrote: Hi, Does Asterisk support wideband VoIP? Is there support for Speex 16 KHz? What devices support this? As I'm sure you know, if the call hits the PSTN it will always be ulaw or alaw and so will not be wideband. Wideband codecs are only helpful if the call IP end to end. ___ --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] Purposely setting red alarm on PRI for testing purposes
On Wed, 11 Apr 2007, Eric Bishop said something to this effect: Does anyone know if is possible to purposely set red alarm status on PRI circuit for testing purposes (other than unplugging it). I have looked for a console command which might allow this That depends on the equipment into which it is terminated. :-) If the T1 dumps into an offboard CSU/DSU, you can loop back the network side toward the telco which will cause an alarm to be perceived on the DTE side. Most T1 cards and T1 interfaces with integrated CSU/DSUs will allow you to do something similar for testing, which will have the effect of putting the interface into the same state it would be in if you unplugged it. Are you bringing this PRI into a Digium PRI card? -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --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] Nagios asterisk monitoring
Hi,
You might have to be a little more specific about what you mean when you
say that it doesn't run from the web interface. Also, such errors might
show up in /var/log/nagios.log. But all other things being equal, it
sounds like it might be an execution permissions issue.
More information would help!
Thanks,
-- Alex
On Wed, 11 Apr 2007, voip crazy said something to this effect:
Dear list,
I am trying to configure the nagios plugin called check_sip. I just read the
README file included with the plugin. I follow the readme steps to configure
the plugin, without success. In the nagios web interface I can see (No
output!) In the status information column. If I run the chech_sip plugin
from a linux console, I get
/usr/local/nagios/libexec# ./check_sip -u sip:[EMAIL PROTECTED]
SIP 200 OK: 0.00 second response time
I do not know why If I run the plugin from the consle it works ok, but if I
run it from Nagios web interface it does not run.
Anyone are using this plugin?
Could you helpme to solve that?
Any clue will be appreciated.
Thanks for your time.
VoipCrazy
Here goes my nagios check_sip plugin configuration.
define command{
command_namecheck_sip
command_line$USER1$/check_sip -u $ARG1$ -H $HOSTADDRESS$ -w 5
}
define service{
use generic-service
host_name -PBX
service_description SIP test
check_command check_sip!sip:[EMAIL PROTECTED]
contact_groups admins
max_check_attempts 4
normal_check_interval 5
retry_check_interval1
notification_interval 240
check_period24x7
notification_period 24x7
notification_optionsc,r
}
--
Alex Balashov [EMAIL PROTECTED]
___
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Re: [asterisk-users] Verizon-Vonage Lawsuit
Christopher Chan wrote: Thanks for the answer. I've never heard that one before. I remember once I used a 3rd party component set (Indy 9) to do some smtp alert emails from a Windows application. Couldn't get the mails to go through for a few particular customers and after some research and talking to ISP's, we found out that the component set that I used was used to write spamming software and the headers produced by the component set were flagged by spamm assasin and others. Arg. Mail admins usually treat non-MTA software that send email as highly potential sources of spam. In your case, a whitelisting arrangement would take care of the problem or whatever arrangement they may have. We simply switched component sets to fix the problem. Pity since Indy is a Wonderful suite of components. Properly managed ISP's/domains will have a reachable contact at [EMAIL PROTECTED]/domain. The third party component set you mentioned...is it the original of indyproject.org too? Yeah, Indy 9 from Indy Project. I've since moved over to Synapse because I am doing a lot of work with Lazarus/Freepascal now and Synapse works great with both Delphi and Freepascal. The Indy 9 component most probably got itself into blacklists because of the IntraWeb framework, which most probably uses Indy 9 for smtp, has inadequate measures against abuse and the Indy 9 component creates an easily identifiable header. I'm not sure that Intraweb has an included smtp interface, but I am almost certain Indy was used as the underlying technology or at least the basis of. I have a Intraweb license (v. 6/7?) actually, but haven't used it much since I started with ASP.net in .net/mono and C#/Chrome. Personally, I'm excited about Delphi for PHP coming out. But now I stray off topic... -- Warm Regards, Lee ___ --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] wideband codec support
Yes. I understand that. I am considering a case where the call is IP end to end. Madhuri --- Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: Madhuri Patwardhan wrote: Hi, Does Asterisk support wideband VoIP? Is there support for Speex 16 KHz? What devices support this? As I'm sure you know, if the call hits the PSTN it will always be ulaw or alaw and so will not be wideband. Wideband codecs are only helpful if the call IP end to end. ___ --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 We won't tell. Get more on shows you hate to love (and love to hate): Yahoo! TV's Guilty Pleasures list. http://tv.yahoo.com/collections/265 ___ --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] calls bridging
Yes. See: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out If you have trouble implementing it from that let me know. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Pezhman Lali Sent: Wednesday, April 11, 2007 7:08 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] calls bridging dear can asterisk dial two numbers, then bridge them.(like jah jah) best Mani __ __ Looking for earth-friendly autos? Browse Top Cars by Green Rating at Yahoo! Autos' Green Center. http://autos.yahoo.com/green_center/ ___ --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] Execute EAGI script with params from extensions.conf
How can I execute an EAGI script with params from extensions.conf Example python script: InfMsg -s 1 in my extensions.conf exten = 492,1,Answer exten = 492,2,eagi,InfMsg -s 1 exten = 492,3,Hangup() It doesn´t work my * report... -- Executing [EMAIL PROTECTED]:2] EAGI(Zap/4-1, InfMsg -s 1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/InfMsg -s 1 == InfoTerminal -s 1: Failed to execute '/var/lib/asterisk/agi-bin/InfMsg -s 1': No such file or directory ___ --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] Execute EAGI script with params fromextensions.conf
I don't think you can put arguments to the agi. Try is as: exten = 492,2,eagi,InfMsg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of equis software Sent: Wednesday, April 11, 2007 10:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Execute EAGI script with params fromextensions.conf How can I execute an EAGI script with params from extensions.conf Example python script: InfMsg -s 1 in my extensions.conf exten = 492,1,Answer exten = 492,2,eagi,InfMsg -s 1 exten = 492,3,Hangup() It doesn´t work my * report... -- Executing [EMAIL PROTECTED]:2] EAGI(Zap/4-1, InfMsg -s 1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/InfMsg -s 1 == InfoTerminal -s 1: Failed to execute '/var/lib/asterisk/agi-bin/InfMsg -s 1': No such file or directory ___ --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] Execute EAGI script with params from extensions.conf
You need to pass the arguments as separate arguments to the EAGI dialplan application, eg: exten = 492,2,EAGI(InfMsg,-s,1) but I would recommend using pipes... exten = 492,2,EAGI(InfMsg|-s|1) But maybe that's just me. On 4/11/07, equis software [EMAIL PROTECTED] wrote: How can I execute an EAGI script with params from extensions.conf Example python script: InfMsg -s 1 in my extensions.conf exten = 492,1,Answer exten = 492,2,eagi,InfMsg -s 1 exten = 492,3,Hangup() It doesn´t work my * report... -- Executing [EMAIL PROTECTED]:2] EAGI(Zap/4-1, InfMsg -s 1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/InfMsg -s 1 == InfoTerminal -s 1: Failed to execute '/var/lib/asterisk/agi-bin/InfMsg -s 1': No such file or directory ___ --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] Execute EAGI script with params from extensions.conf
equis software wrote: How can I execute an EAGI script with params from extensions.conf Example python script: InfMsg -s 1 in my extensions.conf exten = 492,1,Answer exten = 492,2,eagi,InfMsg -s 1 exten = 492,3,Hangup() It doesn´t work my * report... -- Executing [EMAIL PROTECTED]:2] EAGI(Zap/4-1, InfMsg -s 1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/InfMsg -s 1 == InfoTerminal -s 1: Failed to execute '/var/lib/asterisk/agi-bin/InfMsg -s 1': No such file or directory Commas. exten = 492,2,eagi(InfMsg,-s,1) At leeast in older versions of Asterisk, AGI only supports 1 parameter. I got around this by separating options with an . Then break them out into options within my script. I don't know if 1.2 or 1.4 fixed this. ___ --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] Execute EAGI script with params fromextensions.conf
Griepentrog Scott wrote: I don't think you can put arguments to the agi. Try is as: exten = 492,2,eagi,InfMsg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of equis software Sent: Wednesday, April 11, 2007 10:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Execute EAGI script with params fromextensions.conf How can I execute an EAGI script with params from extensions.conf Example python script: InfMsg -s 1 in my extensions.conf exten = 492,1,Answer exten = 492,2,eagi,InfMsg -s 1 exten = 492,3,Hangup() It doesn´t work my * report... -- Executing [EMAIL PROTECTED]:2] EAGI(Zap/4-1, InfMsg -s 1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/InfMsg -s 1 == InfoTerminal -s 1: Failed to execute '/var/lib/asterisk/agi-bin/InfMsg -s 1': No such file or directory Then why not try EAGI(InfMsg,-s 1) or EAGI(InfMsg,-s,1) ? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Execute EAGI script with params fromextensions.conf
Yes you can. Read: http://www.voip-info.org/wiki-Asterisk+AGI On 4/11/07, Griepentrog Scott [EMAIL PROTECTED] wrote: I don't think you can put arguments to the agi. Try is as: exten = 492,2,eagi,InfMsg -Original Message- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] Behalf Of *equis software *Sent:* Wednesday, April 11, 2007 10:54 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Execute EAGI script with params fromextensions.conf How can I execute an EAGI script with params from extensions.conf Example python script: InfMsg -s 1 in my extensions.conf exten = 492,1,Answer exten = 492,2,eagi,InfMsg -s 1 exten = 492,3,Hangup() It doesn´t work my * report... -- Executing [EMAIL PROTECTED]:2] EAGI(Zap/4-1, InfMsg -s 1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/InfMsg -s 1 == InfoTerminal -s 1: Failed to execute '/var/lib/asterisk/agi-bin/InfMsg -s 1': No such file or directory ___ --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] GTalk and No Audio Problem
Hi, i've been trying to connect Asterisk with Google Talk such as some others have tried. Therefor i followed the instructions on http://www.voip-info.org/wiki/view/Asterisk+Google+Talk I took the latest version of asterisk from trunk. My Asterisk server is not NATed but the Google Talk Client is. Signalling a call is no problem, but after the call is set up, no audio is passed. I can see a lot of STUN output such as: STUN Packet, msg Binding Response (0101), length: 48 Found STUN Attribute Username (0006), length 32 Found STUN Attribute Mapped Address (0001), length 8 Ignoring STUN attribute Mapped Address (0001), length 8 Dunno what to do with STUN message 0101 (Binding Response) and no RTP packets are arriving at my client. I have read these bugs that seem to be related: 8193 9401 7686 on http://bugs.digium.com i have tried the various patches that claimed to solve the NAT/STUN problem, so that the rtp packages are routed correctly. What i want to know is the following. I have seen one or two messages from people who claim to have sccessfully audio between asterisk and an gtalk client. It would be very nice, if one of them could share some information how exactly they have done it and with what setup. Don't let us stay in the rain ;) Thanks for you help, Tobias Wolf ___ --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] SIP INFO message
I've got a very strange problem and I can't figure it out. I have a Cisco PRI gateway connected to * via SIP. When I debug on the Cisco, I see callerID name, but it is not getting to * via SIP. I am running * 1.4.2 and the latest Cisco IOS for my router. Here is what is happening: A call comes into the gateway. It sends a SIP INVITE to * with pending as the callerID name (this does NOT show up on any phones). * sends a TRYING message back to the gateway. * waits 2 seconds (I have a 2 second wait in the dialplan) and then sends an INVITE to the phone. The phone sends back TRYING and RINGING to *. * then sends RINGING to the gateway and the gateways sends a SIP INFO with the correct CALLERID NAME. It doesn't matter if the wait in the dialplan is 1 second, 2 seconds, or 5 seconds, it never sends the correct name until after * sends it a RINGING message. I never see any name on my display (neither pending, nor the real name). I am grabbing a tcpdump and I see pending and the real name in there, I just never see it on the * console, or on the phone. The config on * for the gateway is pretty vanilla: [192.168.1.100] context=default type=friend host=192.168.1.100 dtmfmode=rfc2833 disallow=all allow=ulaw allow=g729 canreinvite=yes qualify=yes t38pt_udptl = yes * doesn't appear to understand the INFO message as it is spitting out some errors like below, and I am dropping calls after ~ 30 seconds. [Apr 9 22:52:22] WARNING[14660]: chan_sip.c:1916 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. If I disable the feature on the gateway that sends the name, everything works fine, but I obviously don't get name. I've spoken to several other people that have virtually the same gateway config as me and theirs works. I've tried this with * 1.4.2 and 1.0.3 and I get the same results on both of them. I am to the point where I think I have some * config wrong, but I can't imagine what it could be. Anybody have any insight into why * would freak out on an INFO message? I can send Ethereal captures if that would help. Thanks. Peder ___ --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] Execute EAGI script with params fromextensions.conf
Thanks! At least in asterisk 1.4 exten = 492,2,EAGI(InfMsg|-s|1) work very well On 4/11/07, Philipp Kempgen [EMAIL PROTECTED] wrote: Griepentrog Scott wrote: I don't think you can put arguments to the agi. Try is as: exten = 492,2,eagi,InfMsg -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] Behalf Of equis software Sent: Wednesday, April 11, 2007 10:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Execute EAGI script with params fromextensions.conf How can I execute an EAGI script with params from extensions.conf Example python script: InfMsg -s 1 in my extensions.conf exten = 492,1,Answer exten = 492,2,eagi,InfMsg -s 1 exten = 492,3,Hangup() It doesn´t work my * report... -- Executing [EMAIL PROTECTED]:2] EAGI(Zap/4-1, InfMsg -s 1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/InfMsg -s 1 == InfoTerminal -s 1: Failed to execute '/var/lib/asterisk/agi-bin/InfMsg -s 1': No such file or directory Then why not try EAGI(InfMsg,-s 1) or EAGI(InfMsg,-s,1) ? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Cisco IP Phone services.xml sample?
Does anyone have a small, plain services.xml file for a cisco ip phone, preferably one that will work on a 7960? I can't seem to get my xml right, and no matter what I send to the phone I keep getting parse errors. Thanks Shawn ___ --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] SIP INFO message
It sounds as if the gateway is configured to propogate caller ID number resolution and CNAM via SIP INFO, while Asterisk expects to receive it in the contact information associated with the INVITE transaction (i.e. From:). I think the main motivation for doing it this way from the gateway's standpoint is that CNAM dips can sometimes take additional time, so it's better to ensure there's a way to send them as a midsession signaling addition instead of with the initial ring. But that's just a complete off-the-cuff guess. I do not know how to make Asterisk get caller ID that way instead, but I will investigate and see if I come up with anything. -- Alex On Wed, 11 Apr 2007, Peder @ NetworkOblivion said something to this effect: I've got a very strange problem and I can't figure it out. I have a Cisco PRI gateway connected to * via SIP. When I debug on the Cisco, I see callerID name, but it is not getting to * via SIP. I am running * 1.4.2 and the latest Cisco IOS for my router. Here is what is happening: A call comes into the gateway. It sends a SIP INVITE to * with pending as the callerID name (this does NOT show up on any phones). * sends a TRYING message back to the gateway. * waits 2 seconds (I have a 2 second wait in the dialplan) and then sends an INVITE to the phone. The phone sends back TRYING and RINGING to *. * then sends RINGING to the gateway and the gateways sends a SIP INFO with the correct CALLERID NAME. It doesn't matter if the wait in the dialplan is 1 second, 2 seconds, or 5 seconds, it never sends the correct name until after * sends it a RINGING message. I never see any name on my display (neither pending, nor the real name). I am grabbing a tcpdump and I see pending and the real name in there, I just never see it on the * console, or on the phone. The config on * for the gateway is pretty vanilla: [192.168.1.100] context=default type=friend host=192.168.1.100 dtmfmode=rfc2833 disallow=all allow=ulaw allow=g729 canreinvite=yes qualify=yes t38pt_udptl = yes * doesn't appear to understand the INFO message as it is spitting out some errors like below, and I am dropping calls after ~ 30 seconds. [Apr 9 22:52:22] WARNING[14660]: chan_sip.c:1916 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. If I disable the feature on the gateway that sends the name, everything works fine, but I obviously don't get name. I've spoken to several other people that have virtually the same gateway config as me and theirs works. I've tried this with * 1.4.2 and 1.0.3 and I get the same results on both of them. I am to the point where I think I have some * config wrong, but I can't imagine what it could be. Anybody have any insight into why * would freak out on an INFO message? I can send Ethereal captures if that would help. Thanks. Peder ___ --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 -- Alex Balashov [EMAIL PROTECTED] ___ --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] SIP INFO message
Peder @ NetworkOblivion wrote: I've got a very strange problem and I can't figure it out. I have a Cisco PRI gateway connected to * via SIP. When I debug on the Cisco, I see callerID name, but it is not getting to * via SIP. I am running * 1.4.2 and the latest Cisco IOS for my router. Here is what is happening: A call comes into the gateway. It sends a SIP INVITE to * with pending as the callerID name (this does NOT show up on any phones). * sends a TRYING message back to the gateway. * waits 2 seconds (I have a 2 second wait in the dialplan) and then sends an INVITE to the phone. Maybe you could try something like Wait(2) on the *gateway* to wait until the caller-id dropped in over ISDN? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] wideband codec support
Peer 2 Peer SIP (or other protocol) is the future of telephony. Speex will work but you need to compile and install it separately from Asterisk (and before Asterisk) due to licensing issues with Digium. Not sure why you would want to use speex but here is a good link on the wiki with additional info. http://www.voip-info.org/wiki/view/Asterisk+config+codecs.conf I have only used it for very low bandwidth applications and satellite connections. It increased latency but the audio was almost perfect where other codecs broke up so badly that a conversation was impossible. Thanks, Steve Madhuri Patwardhan wrote: Yes. I understand that. I am considering a case where the call is IP end to end. Madhuri --- Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: Madhuri Patwardhan wrote: Hi, Does Asterisk support wideband VoIP? Is there support for Speex 16 KHz? What devices support this? As I'm sure you know, if the call hits the PSTN it will always be ulaw or alaw and so will not be wideband. Wideband codecs are only helpful if the call IP end to end. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: ___ --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 330/320
Honestly, I can't remember and I've seen them ... but should be a bit better. IP 301 phone - 4 line x 20 character monochrome display IP 320/330 - 102 x 33 pixel graphical LCD Not backlit from what I can see still (correct me if I'm wrong). and heads up ... NO POWER SUPPLY'S ARE INCLUDED! (Just trying to prevent people from making a mistake, since this is the first time Polycom will be doing the Cisco approach to the power supply). Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http:// voipstore.atacomm.com/ On Apr 11, 2007, at 1:40 AM, Andrew Joakimsen wrote: How is the screen compared to the other Polycom products? On 4/9/07, Jessee J Holmes [EMAIL PROTECTED] wrote: Mike, I don't have much information, except they are due for shipment soon (mid to end of April to distribution from Polycom). We've demoed a couple and I personally believe they'll be a tough phone to find in stock for the first few months their released. Demand on these from what I'm seeing right now is very, very high. I think they are a great addition to the family and most importantly they have FULL DUPLEX SPEAKERPHONE! :) 550's are released products though. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Apr 9, 2007, at 3:55 PM, Mike wrote: Ah, thanks. I didn't realize this. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Nickerson Sent: Monday, April 09, 2007 16:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 330/320 Mike [EMAIL PROTECTED] wrote: How do you guys like the 330 and 320? Mike, As far as I am aware, neither of these handsets are presently shipping from Polycom, so most people's experience will be limited to PDF brochures and pretty pictures. On the face of it, this looks like a good alternative to the IP301 since it adds native 802.3af PoE support. Not sure yet exactly where the pricing will slot in, however. -Darren -- Darren Nickerson Senior Sales Support Engineer Telephony Depot www.telephonydepot.com +1.215.825.8710 ext 8106 (office) +1.215.243.8335 (fax) ___ --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 ___ --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] Execute EAGI script with paramsfromextensions.conf
duh. thaanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Sean Bright Sent: Wednesday, April 11, 2007 11:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Execute EAGI script with paramsfromextensions.conf Yes you can. Read: http://www.voip-info.org/wiki-Asterisk+AGI On 4/11/07, Griepentrog Scott [EMAIL PROTECTED] wrote: I don't think you can put arguments to the agi. Try is as: exten = 492,2,eagi,InfMsg -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] Behalf Of equis software Sent: Wednesday, April 11, 2007 10:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Execute EAGI script with params fromextensions.conf How can I execute an EAGI script with params from extensions.conf Example python script: InfMsg -s 1 in my extensions.conf exten = 492,1,Answer exten = 492,2,eagi,InfMsg -s 1 exten = 492,3,Hangup() It doesn´t work my * report... -- Executing [EMAIL PROTECTED]:2] EAGI(Zap/4-1, InfMsg -s 1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/InfMsg -s 1 == InfoTerminal -s 1: Failed to execute '/var/lib/asterisk/agi-bin/InfMsg -s 1': No such file or directory ___ --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 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] outCALL- the open source Asterisk integration applicaiton for Microsoft Outlook
Bicom Systems releases outCALL, an Asterisk open source Outlook integration LONDON, UK (11th April 2007) - Bicom Systems announced today it has released outCALL, an open source desktop application allowing integration Microsoft Outlook. OutCALL allows users an easy way for placing and receiving phone calls integrated with users Outlook contacts. The open source PBX market needed integration with Microsoft Outlook which works with Asterisk (www.asterisk.org) . After developing and offering outCALL as a proprietary application, we decided to release outCALL as an open source application licensed under BSD license in order to further stimulate development and use of Asterisk said Senad Jordanovic, the systems architect at Bicom Systems Ltd. We released OutCALL as open source in the wish that the application would to be of good use to everyone and to be enjoyed it for free. Some things are not just about money, we are pleased to contribute to the wider community said Sergej Kasumovic, Chief Developer at Bicom System Ltd. OutCALL is written in C++ . It is a stable and robust application. It took many months of hard work to get it into current state. At the end, I would just say that I will be very happy if OutCALL will make difference to someone. said Denis Komaradic, OutCALL developer at Bicom Systems Ltd. There is ever growing demand to see existing CRM style packages integrated with Telephony Platforms. There are many CRM programs both proprietary and open-source that could benefit from this code. We chose the BSD licence as in our opinion it allows for the broadest possible promotion of the software. We look forward to seeing this open up many more possible integrations with other existing software both by in-house and other commercial vendors, said Stephen Wingfield at Bicom Systems Ltd. For full details on Bicom Systems products please www.bicomsystems.com. To download a copy of OutCall, please visit http://outcall.sf.net/ .Documentation is available at www.bicomsystems.com/docs/outcall/ . About Bicom Systems Bicom Systems is a provider of PBX and soft switch turn key solutions with a presence in the United States and the European Union and supported by a network of resellers across the world. Its solutions allow easy deployment, maintenance and control of a wide range of telephony solutions. The company leads the industry in providing the most integrated, ready to deploy, feature packed Telephony Solutions for creating PBXs and BROADBAND PHONE COMPANIES. For more information about Bicom Systems, please visit www.bicomsystems.com. Outlook is a registered TradeMark of Microsoft Corporation. In no manner should this press release be understood to represent any relationship between Bicom Systems and Microsoft or any endorsement of either company or the products of either company. For more information, please contact: Stephen Wingfield 44-20-7043-3489 [EMAIL PROTECTED] ___ --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] Call Pickup with more than one argument
Dear all, Does Pickup application accept multiple extensions pickup syntax, like the following line? Pickup(extension1extension2...) I've tried it in Asterisk 1.2.17 but it doesn't work. Does it work in Asterisk 1.4 already? Or is any other way in any version of Asterisk that I can use to do the same thing? Thanks, Ricardo. ___ --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] how to install asterisk on redhat ?
I wrote this ages ago. You may want to get more current software than the URL's that are listed. #YUM INSTALLS yum -y install gcc yum -y install kernel-source yum -y install bison yum -y install doxygen yum -y install openssl-devel yum -y install flex yum -y install gcc # WGET DOWNLOADS FROM H6315 / TARBALLS wget http://www.h6315.com/pub/asterisk/asterisk-1.0.6.tar.gz wget http://www.h6315.com/pub/zaptel/zaptel-1.0.6.tar.gz wget http://www.h6315.com/pub/libpri/libpri-1.0.6.tar.gz wget http://www.h6315.com/pub/openssl/openssl-0.9.7f.tar.gz wget http://www.h6315.com/pub/apache/httpd-2.0.53.tar.gz wget http://www.h6315.com/pub/mysql/mysql-standard-4.1.10a-pc-linux-gnu-i686.tar.gz wget http://www.h6315.com/pub/php/php-4.3.10.tar.gz wget http://www.h6315.com/pub/mpg123/mpg123-0.59r.tar.gz # UNTAR EVERYTHING tar -xvzf asterisk*.tar.gz tar -xvzf zaptel*.tar.gz tar -xvzf libpri*.tar.gz tar -xvzf openssl*.tar.gz tar -xvzf httpd*.tar.gz tar -xvzf mysql-*.tar.gz tar -xvzf php*.tar.gz tar -xvzf mpg123*.tar.gz rm -f *.tar.gz rm -f *.rpm # INSTALL OPEN SSL cd /usr/src/openssl* ./config make make test make install # INSTALL APACHE cd /usr/src/httpd-2* ./configure --prefix=/wwwroot --enable-so --enable-rewrite --enable-headers --enable-expires -enable-deflate --with-z --enable-speling --enable-ssl make make install # INSTALL MYSQL cd /usr/src mv mysql* /usr/local cd /usr/local groupadd mysql useradd -g mysql mysql ln -s mysql-standard-4.1.10a-pc-linux-gnu-i686 mysql cd mysql scripts/mysql_install_db --user=mysql chown -R root . chown -R mysql data chgrp -R mysql . cp support-files/mysql.server /etc/init.d chmod +x /etc/init.d/mysql.server ln -s /usr/local/mysql/bin/mysql /usr/bin/mysql # INSTALL PHP cd /usr/src/php* ./configure --prefix=/wwwroot/php --with-apxs2=/wwwroot/bin/apxs --with-config-file-path=/wwwroot/php --with-mysql --enable-gd --with-mysqli=/usr/local/mysql/bin/mysql_config make make install # INSTALL MPG123 cd /usr/src/mpg123* make linux make install # INSTALL ZAPTEL cd /usr/src/zap* perl -pi~ -e 's/# ztdummy/ztdummy/' Makefile make clean make make install # INSTALL LIBPRI cd /usr/src/libp* make make install #INSTALL ASTERISK cd /usr/src/aster* make clean make make install make samples make progdocs - Original Message - From: Malik Mulki (Plant, Feed, Makassar) To: asterisk-users@lists.digium.com Sent: Wednesday, April 11, 2007 4:16 AM Subject: [asterisk-users] how to install asterisk on redhat ? Hiasterisk users... how to install asterisk on redhat ? -- ___ --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] IMAP Voicemail with MS Exchange
Hi there, We're trying to get IMAP voicemail storage working on an MS Exchange server - I would be grateful if anyone who has successfully done this could post the magic soup here, as extensive Google searching has yielded nothing other than tantalizing references to it being done without any specifics. -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp ___ --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] wideband codec support
Thanks for the prompt reply. I had a look at asterisk source file codec/codec_speex.c and it *seems* like 8 khz is hard coded in it. I am interested in wideband codec and as I understand speex (16 khz and 32 khz) are the only open source wide band codecs. Thanks for the wiki link, I will certainly try it. Is 16 khz Speex possible with Asterisk? Has anybody tried it? Thanks, Madhuri --- Steve Totaro [EMAIL PROTECTED] wrote: Peer 2 Peer SIP (or other protocol) is the future of telephony. Speex will work but you need to compile and install it separately from Asterisk (and before Asterisk) due to licensing issues with Digium. Not sure why you would want to use speex but here is a good link on the wiki with additional info. http://www.voip-info.org/wiki/view/Asterisk+config+codecs.conf I have only used it for very low bandwidth applications and satellite connections. It increased latency but the audio was almost perfect where other codecs broke up so badly that a conversation was impossible. Thanks, Steve Madhuri Patwardhan wrote: Yes. I understand that. I am considering a case where the call is IP end to end. Madhuri --- Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: Madhuri Patwardhan wrote: Hi, Does Asterisk support wideband VoIP? Is there support for Speex 16 KHz? What devices support this? As I'm sure you know, if the call hits the PSTN it will always be ulaw or alaw and so will not be wideband. Wideband codecs are only helpful if the call IP end to end. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: ___ --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 Looking for earth-friendly autos? Browse Top Cars by Green Rating at Yahoo! Autos' Green Center. http://autos.yahoo.com/green_center/ ___ --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] RTP to Sound card...
¡Hola! ¡It would be great if anybody could give me a hint! ¿How does the RTP Data come to the Soundcard? To specify a bit my question: 1. I have an external RTP Stream wich is not part of the Sip-Communicator. Its hard to explain because... 2. Now I want to 'hear' this stream... 3. The Sip-Communcator does it with it own RTP Stream, so it should be able to do it with an external It would be already a big help if anybody gives me a hint what is the coupeling class between RTP and the Soundcard, like 'class X consumes Stream Y and somehow passes it to the sondcard'. So I could create with my RTP the Stream Y and pass it to the class... Mchas Gracias, Laura ___ --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] SIP Jitter Buffer Patch for 1.2.x branch?
Hi, I know that there was a jitter buffer patch (for sip) for the 1.0.9 branch some time agin. At this time, we can not upgrade to 1.4.x. Is there a useable, fairly stable INCOMING sip jitter buffer patch? That is.. I want Asterisk to jitter buffer incoming SIP packets. ___ --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] outCALL- the open source Asterisk integrationapplicaiton for Microsoft Outlook
To download a copy of OutCall, please visit http://outcall.sf.net/ .Documentation is available at www.bicomsystems.com/docs/outcall/ . I forgot to include in my original post: Please use http://sf.net for any further communications in regards to outCALL Thanks Senad ___ --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] Cisco IP Phone services.xml sample?
Does anyone have a small, plain services.xml file for a cisco ip phone, preferably one that will work on a 7960? I can't seem to get my xml right, and no matter what I send to the phone I keep getting parse errors. Thanks Shawn CiscoIPPhoneMenu TitleXML Portal/Title PromptChoose from a range of XML Services:/Prompt MenuItem NameBerbee XML Main Menu/Name URLhttp://phone-xml.berbee.com/menu.xml/URL /MenuItem MenuItem NameBT Exact XML Main Menu/Name URLhttp://193.113.58.136/bt//URL /MenuItem MenuItem NameStock Quotes/Name URLhttp://phone-xml.berbee.com/cgi-bin/stockchk.pl/URL /MenuItem MenuItem NameUS Weather/Name URLhttp://phone-xml.berbee.com/cgi-bin/weather.pl/URL /MenuItem MenuItem NameUK Weather/Name URLhttp://193.113.58.136/bt/weather/weatherinfo.asp/URL /MenuItem MenuItem NamePhil's XML Development Page/Name URLhttp://flame.tiefighter.org/fwd/xml/dev//URL /MenuItem /CiscoIPPhoneMenu ___ --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] Polycom 501 issue with latest firmware : sluggish keys
Hi, I've upgraded a few Polycom 501 to SIP 2.0.3b (can't get 2.1.0 because of Polycom's firmware policy, but this is the latest publicly available from Polycom's web site). I've noticed that some keys get sticky though. Soft buttons for example (i.e. end call) need to be pressed 2-3 times for them to react. I've downgraded to 1.6.7, and the problem dissapeared. I can't imagine I'm the only one having that issue, and that issue was also present in 2.0.1 for me. Did anybody else have this problem? What did you do to fix it? Am I stuck with 1.6.7 forever? Mike ___ --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] Mediatrix 1204
Hi - I've recently bought a mediatrix 1204 and have had a complete nightmare getting it up and running with an [EMAIL PROTECTED] setup. I know this isn't a mediatrix list but I'm at my wits end and the support with this product is atrocious. (mine was even shipped with firmware that was incompatible with the win32 software it came with so I wasted a day trying to work out why the SNMP software wouldn't work ) I've finally managed to get incoming calls to work properly by getting it to forward all calls to 4000 which is then passed on to the asterisk proxy and treated as an inbound route that gets answered correctly. The problem is then that when I place an outbound call through the gateway it also forwards that back as well. It then uses each channel in order until it fails as they're all busy. The xml configuration file is at http://www.ascensus.co.uk/config.xml The asterisk debug log is as below with my mobile replaced with mymobileno: I've also attached sip.conf below. If anyone has any idea how to get this thing to accept outgoing calls I would be very grateful of any input. All the docs and howto's I've found state that it should 'just work' once the inbound settings are working but I've not found that to be the case. The settings are all defaults except the following: Static IP address Proxy server address VAD on 711 disabled Comfort noise disabled AutomaticCallEnable yes AutomaticCallTargetAddress 4000 (which is obviously the problem...) Any help appreciated Thanks robbie Sip.conf snip [inbound] type=friend host=192.168.0.253 context=from-pstn canreinvite=no allow=ulaw allow=alaw snip asterisk1*CLI -- Executing Macro(SIP/4005-9d61, dialout-trunk|7|mymobilenumber|) in new stack -- Executing GotoIf(SIP/4005-9d61, 1?3:2)) in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro(SIP/4005-9d61, record-enable|4005|OUT) in new stack -- Executing GotoIf(SIP/4005-9d61, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing GotoIf(SIP/4005-9d61, 1?5:8) in new stack -- Goto (macro-record-enable,s,5) -- Executing DBget(SIP/4005-9d61, RecEnable=RECORD-OUT/4005) in new stack -- DBget: varname=RecEnable, family=RECORD-OUT, key=4005 -- DBget: Value not found in database. -- Executing SetVar(SIP/4005-9d61, CALLFILENAME=OUT4005-20070411-181258-1176311578.13302) in new stack -- Executing Goto(SIP/4005-9d61, s|14) in new stack -- Goto (macro-record-enable,s,14) -- Executing GotoIf(SIP/4005-9d61, 0?15:99) in new stack -- Goto (macro-record-enable,s,99) -- Executing NoOp(SIP/4005-9d61, NO RECORDING NEEDED) in new stack -- Executing GotoIf(SIP/4005-9d61, fooBgate:?7) in new stack -- Executing SetCallerID(SIP/4005-9d61, Bgate: Treatment (Large) 4005) in new stack -- Executing Goto(SIP/4005-9d61, 9) in new stack -- Goto (macro-dialout-trunk,s,9) -- Executing SetGroup(SIP/4005-9d61, OUT_7) in new stack -- Executing CheckGroup(SIP/4005-9d61, ) in new stack -- Executing SetVar(SIP/4005-9d61, DIAL_NUMBER=mymobilenumber) in new stack -- Executing SetVar(SIP/4005-9d61, DIAL_TRUNK=7) in new stack -- Executing AGI(SIP/4005-9d61, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar(SIP/4005-9d61, OUTNUM=mymobilenumber) in new stack -- Executing Cut(SIP/4005-9d61, custom=OUT_7|:|1) in new stack -- Executing GotoIf(SIP/4005-9d61, 0?19) in new stack -- Executing Dial(SIP/4005-9d61, SIP/inbound/mymobilenumber) in new stack We're at 192.168.0.254 port 12542 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK6728b435 From: Bgate: Treatment (Large) sip:[EMAIL PROTECTED];tag=as5b17ec6a To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 11 Apr 2007 17:12:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 240 v=0 o=root 1321 1321 IN IP4 192.168.0.254 s=session c=IN IP4 192.168.0.254 t=0 0 m=audio 12542 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 192.168.0.253:5060 -- Called inbound/mymobilenumber asterisk1*CLI Sip read: SIP/2.0 100 Trying Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE From: Bgate: Treatment (Large) sip:[EMAIL PROTECTED];tag=as5b17ec6a To: sip:[EMAIL PROTECTED];tag=2120bdca0a07567 Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK6728b435 Content-Length: 0 User-Agent: MxSipApp/5.0.15.88 MxSF/v3.2.1.1 8 headers, 0 lines asterisk1*CLI Sip read: INVITE sip:[EMAIL PROTECTED] SIP
Re: [asterisk-users] IMAP Voicemail with MS Exchange
Interesting. I've generally only seen this done with unified messaging systems, like Cisco Unity, and some proprietary ones I've worked with, which involve taking messages of various media types (fax, voice, e-mail) and storing them using some intermediate abstraction/middleware and then designing access services which talk to said layer, be it a POP or IMAP server. -- Alex Balashov [EMAIL PROTECTED] ___ --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] SIP Jitter Buffer Patch for 1.2.x branch?
On Wed, 11 Apr 2007, Matt said something to this effect: I know that there was a jitter buffer patch (for sip) for the 1.0.9 branch some time agin. At this time, we can not upgrade to 1.4.x. Is there a useable, fairly stable INCOMING sip jitter buffer patch? That is.. I want Asterisk to jitter buffer incoming SIP packets. Incoming RTP packets you mean? :-) I am not aware that a jitter buffer patch exists for 1.0.x. But I could be wrong; however, when I ran into this exact same issue I was not able to find anything. But I didn't try very hard. This is a question you may possibly want to ask on the asterisk-dev list. -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --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] SIP Jitter Buffer Patch for 1.2.x branch?
On Wed, 2007-04-11 at 13:15 -0400, Matt wrote: Hi, I know that there was a jitter buffer patch (for sip) for the 1.0.9 branch some time agin. At this time, we can not upgrade to 1.4.x. Is there a useable, fairly stable INCOMING sip jitter buffer patch? That is.. I want Asterisk to jitter buffer incoming SIP packets. Maybe this one? http://asterisk-backports.org/wiki/index.php/Rtp-jb-1.2 Regards, Patrick ___ --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] Verizon-Vonage Lawsuit
Salvatore Giudice wrote: BTW, the main problem with these patents is that they tend to lower the rate of adoption for new standards. Nothing kills a standard quicker than when someone patents it. For example, someone out there even has a patent on ENUM: http://www.freepatentsonline.com/20060020713.html?highlight=enumstemming=on It made me mad that he beat me to it. Roflol... Regardless, this won't help with ENUM adoption. Any joker with about $6k per patent and some time on his hands to monitor emerging standards can easily generate some patent entertainment for themselves at the expense of others... So, the question of the day is: Have you thought about patenting something today? It's easy. I just got a new idea while writing this for an ENUM related patent that I may pursue at some point... =) The US patent system is totally broken. It started with lobbying efforts to relax the applicability rules for patents for short-term gain. In the long term, it's going to do big damage to American competitiveness. And that's the sad thing about this. It discourages actual innovation (despite Wall Street protests to the contrary). If everytime you want to build on somebody else's work you have to build a skein of licencing agreements, you start to ask yourself, why should I bother? More and more companies are answering that one with We shouldn't -- there's enough action to be had in other parts of the world, where the conditions are much less onerous. Another example of that kind of short-sighted thinking is what happened to the US crypto business when all the export controls were brought in. (A lot of damage was done in exchange for no demonstrable security benefit.) Obviously, a market that big and moneyed isn't going to be ignored: how can it be? But what used to be a no-brainer isn't so obvious anymore -- staying out of the US market is a serious option where it wasn't before, and that just leads to further Balkanization. It's fitting that an open source product like Asterisk is helping keep the US in the game. -Stephen- ___ --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] Play audio and continue to next priority before audio ends...
Am Monday 09 April 2007 23:20 schrieb Alejandro Mejía: Hello list members. I would like to know how to playback an audio file to the caller, and while it's played asterisk to continue executing the next priorities on extensions.conf That's not the case when using playback command, because the next priority is executed until the audio file ends playing. I want to evaluate some variables while caller hears the audio file. Any ideas? Use StartMusicOnHold() and StopMusicOnHold() If use use MoH type=files MoH play from start of file, so ist similar to Playback ___ --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] IMAP Voicemail with MS Exchange
Anthony Rodgers wrote: Hi there, We're trying to get IMAP voicemail storage working on an MS Exchange server - I would be grateful if anyone who has successfully done this could post the magic soup here, as extensive Google searching has yielded nothing other than tantalizing references to it being done without any specifics. I haven't used IMAP voicemail yet, so you'll have to bear with me here. Have you tried configuring Asterisk to save voicemail messages on the Exchange server using IMAP? What was the result? IMAP support in Exchange, as in Outlook, is rough and rather ugly. For obvious reasons it's never been in MS interest to support it properly, as they want people to use their native Exchange server protocol. There's probably a good reason why you want to do it the way you want to, but I'll ask the question anyway -- what about delivering the message over SMTP? -Stephen- ___ --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] how to install asterisk on redhat ?
On Wed, Apr 11, 2007 at 07:32:52PM +0300, Dovid B wrote: I wrote this ages ago. You may want to get more current software than the URL's that are listed. #YUM INSTALLS yum -y install gcc yum -y install kernel-source actually: kernel-devel (or kernel-smp-devel) yum -y install bison yum -y install doxygen yum -y install openssl-devel yum -y install flex yum -y install gcc # WGET DOWNLOADS FROM H6315 / TARBALLS wget http://www.h6315.com/pub/asterisk/asterisk-1.0.6.tar.gz wget http://www.h6315.com/pub/zaptel/zaptel-1.0.6.tar.gz wget http://www.h6315.com/pub/libpri/libpri-1.0.6.tar.gz wget http://www.h6315.com/pub/openssl/openssl-0.9.7f.tar.gz wget http://www.h6315.com/pub/apache/httpd-2.0.53.tar.gz wget http://www.h6315.com/pub/mysql/mysql-standard-4.1.10a-pc-linux-gnu-i686.tar.gz wget http://www.h6315.com/pub/php/php-4.3.10.tar.gz wget http://www.h6315.com/pub/mpg123/mpg123-0.59r.tar.gz What is the point in rebuilding stuff that is already availble from your distribution? And is actively mintained by it? I hope whoever installed by such a tutorial is not still using those obsolete versions. # UNTAR EVERYTHING tar -xvzf asterisk*.tar.gz tar -xvzf zaptel*.tar.gz tar -xvzf libpri*.tar.gz tar -xvzf openssl*.tar.gz tar -xvzf httpd*.tar.gz tar -xvzf mysql-*.tar.gz tar -xvzf php*.tar.gz tar -xvzf mpg123*.tar.gz rm -f *.tar.gz rm -f *.rpm # INSTALL OPEN SSL cd /usr/src/openssl* ./config make make test make install # INSTALL APACHE cd /usr/src/httpd-2* ./configure --prefix=/wwwroot --enable-so --enable-rewrite --enable-headers --enable-expires -enable-deflate --with-z --enable-speling --enable-ssl make make install # INSTALL MYSQL cd /usr/src mv mysql* /usr/local cd /usr/local groupadd mysql useradd -g mysql mysql ln -s mysql-standard-4.1.10a-pc-linux-gnu-i686 mysql cd mysql scripts/mysql_install_db --user=mysql chown -R root . chown -R mysql data chgrp -R mysql . cp support-files/mysql.server /etc/init.d chmod +x /etc/init.d/mysql.server ln -s /usr/local/mysql/bin/mysql /usr/bin/mysql # INSTALL PHP cd /usr/src/php* ./configure --prefix=/wwwroot/php --with-apxs2=/wwwroot/bin/apxs --with-config-file-path=/wwwroot/php --with-mysql --enable-gd --with-mysqli=/usr/local/mysql/bin/mysql_config make make install # INSTALL MPG123 cd /usr/src/mpg123* make linux make install # INSTALL ZAPTEL cd /usr/src/zap* perl -pi~ -e 's/# ztdummy/ztdummy/' Makefile make clean make make install # INSTALL LIBPRI cd /usr/src/libp* make make install #INSTALL ASTERISK cd /usr/src/aster* make clean make make install make samples make progdocs Have some mercy on the CPU and HD, and spare this one... -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Call Pickup with more than one argument
On Wed, Apr 11, 2007 at 05:27:51PM +0100, Ricardo Carvalho wrote: Dear all, Does Pickup application accept multiple extensions pickup syntax, like the following line? Pickup(extension1extension2...) I've tried it in Asterisk 1.2.17 but it doesn't work. Does it work in Asterisk 1.4 already? Or is any other way in any version of Asterisk that I can use to do the same thing? I believe that the Bristuff ChanPickup supports this. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] HPEC audio clipping
Greg Siemon wrote: Thanks for the helps Stephen. I was running non standard gains but setting regain and txgain to zero (then reloading chan_zap.so) does not help. I still get the broken audio, in fact sometimes I don't get any audio at all. In testing the server just froze a number of times and had to be rebooted via the power switch. I am using the latest Zaptel 1.2.16 files and the latest fxotune from the 1.4 release and I still see this issue. Very interested to get this working but without the HPEC my server is rock solid (only have to reboot it when I install kernel updates). I don't believe it is my system but am happy to do any testing others may suggest. Have you had any luck with this, Greg? -Stephen- ___ --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] SIP Jitter Buffer Patch for 1.2.x branch?
I did find the jitter buffer patch on the bug-tracker...(ast_jb-1.2.0.patch4). I applied it to a 1.2.6 asterisk and it seemed to apply all but 2 small chunks (which I was able to apply myself)... it then compiled... so I'm going to give it a shot and test it out. I will report back results. On 4/11/07, Alex Balashov [EMAIL PROTECTED] wrote: On Wed, 11 Apr 2007, Matt said something to this effect: I know that there was a jitter buffer patch (for sip) for the 1.0.9branch some time agin. At this time, we can not upgrade to 1.4.x. Is there a useable, fairly stable INCOMING sip jitter buffer patch? That is.. I want Asterisk to jitter buffer incoming SIP packets. Incoming RTP packets you mean? :-) I am not aware that a jitter buffer patch exists for 1.0.x. But I could be wrong; however, when I ran into this exact same issue I was not able to find anything. But I didn't try very hard. This is a question you may possibly want to ask on the asterisk-dev list. -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --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] Nagios asterisk monitoring
I wrote a very extensive plugin for cacti to monitor asterisk.
It uses the manager interface to poll and get statistics for 1.4 and 1.2.
Let me know if you interested, ill post it, or email me directly.
-bkruse
voip crazy wrote:
Dear list,
I am trying to configure the nagios plugin called check_sip. I just
read the README file included with the plugin. I follow the readme
steps to configure the plugin, without success. In the nagios web
interface I can see (No output!) In the status information column. If
I run the chech_sip plugin from a linux console, I get
/usr/local/nagios/libexec# ./check_sip -u sip:[EMAIL PROTECTED]
SIP 200 OK: 0.00 second response time
I do not know why If I run the plugin from the consle it works ok, but
if I run it from Nagios web interface it does not run.
Anyone are using this plugin?
Could you helpme to solve that?
Any clue will be appreciated.
Thanks for your time.
VoipCrazy
Here goes my nagios check_sip plugin configuration.
define command{
command_namecheck_sip
command_line$USER1$/check_sip -u $ARG1$ -H $HOSTADDRESS$ -w 5
}
define service{
use generic-service
host_name -PBX
service_description SIP test
check_command check_sip!sip:[EMAIL PROTECTED]
contact_groups admins
max_check_attempts 4
normal_check_interval 5
retry_check_interval1
notification_interval 240
check_period24x7
notification_period 24x7
notification_optionsc,r
}
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Re: [asterisk-users] Re: incoming zaptel calls fail
Try to update your zaptel to latest 1.4 svn. I just fixed a bug in a patch that was committed not too long ago. It should fix it. Matthew Fredrickson On Apr 9, 2007, at 7:06 PM, Robert La Ferla wrote: On Apr 9, 2007, at 1:51 PM, Kevin P. Fleming [EMAIL PROTECTED] wrote: You also neglected to mention the version of Asterisk you are running; 'latest SVN' means nothing when there are 20+ branches of Asterisk on our SVN server. Sorry about that. It is the 1.4 trunk: Asterisk SVN-branch-1.4-r60850, Copyright (C) 1999 - 2006 Digium, Inc. and others. and to recap: OS: Fedora Core 5: Linux myhost 2.6.20-1.2307.fc5 #1 Sun Mar 18 20:44:48 EDT 2007 i686 i686 i386 GNU/Linux ZAPTEL: Zaptel Version: SVN-branch-1.4-r2397M ERROR MSGS: [Apr 9 09:38:02] WARNING[16564]: chan_zap.c:6470 ss_thread: CallerID returned with error on channel 'Zap/2-1' [Apr 9 09:38:50] WARNING[16570]: chan_zap.c:6470 ss_thread: CallerID returned with error on channel 'Zap/2-1' [Apr 9 09:39:34] WARNING[16580]: chan_zap.c:6470 ss_thread: CallerID returned with error on channel 'Zap/2-1' ___ --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] SIP Jitter Buffer Patch for 1.2.x branch?
Hey that looks like it might do it! On 4/11/07, Patrick [EMAIL PROTECTED] wrote: On Wed, 2007-04-11 at 13:15 -0400, Matt wrote: Hi, I know that there was a jitter buffer patch (for sip) for the 1.0.9 branch some time agin. At this time, we can not upgrade to 1.4.x. Is there a useable, fairly stable INCOMING sip jitter buffer patch? That is.. I want Asterisk to jitter buffer incoming SIP packets. Maybe this one? http://asterisk-backports.org/wiki/index.php/Rtp-jb-1.2 Regards, Patrick ___ --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] Polycom - Static IP
Is there a way to config a static ip address on a Polycom phone remotely ie. From a config file or a web browser? Steve ___ --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] missing chan_zap.so
Few days back I installed Asterisk 1.4.2 with Zaptel 1.4.0. All SIP accounts were working fine, today I tried to install a fxs Sangoma A200 card and got the following error. [Apr 12 01:15:17] WARNING[31018]: channel.c:3024 ast_request: No channel type registered for 'Zap' [Apr 12 01:15:17] WARNING[31018]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) Searched google and came to conclusion that I was missing chan_zap.so on my machine. Followed the instruction of the bug at http://bugzilla.atrpms.net/show_bug.cgi?id=1165 and downloaded zaptel 1.4.1, after that executed the following commands ./configure make clean make make install Went to asterisk folder ./configure make clean make make upgrade But could not get chan_zap.so then did the make install of asterisk. still missing the chan_zap.so Can someone please help. Regards, Sanjay Rajdev ___ --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 501 issue with latest firmware : sluggish keys
I've seen an issue like this from time to time on 601s, even with the latest firmware. Not just the softkeys, but also the dial keys. The phones seem to run slow sometimes, failing to respond to a key press right away but getting to it eventually. It usually clears up after a few seconds. Also, I've noticed that the 601s sometimes ignore key presses altogether, just as you describe. I have not yet found a solution for this problem... On Apr 11, 2007, at 1:37 PM, Mike wrote: Hi, I've upgraded a few Polycom 501 to SIP 2.0.3b (can't get 2.1.0 because of Polycom's firmware policy, but this is the latest publicly available from Polycom's web site). I've noticed that some keys get sticky though. Soft buttons for example (i.e. end call) need to be pressed 2-3 times for them to react. I've downgraded to 1.6.7, and the problem dissapeared. I can't imagine I'm the only one having that issue, and that issue was also present in 2.0.1 for me. Did anybody else have this problem? What did you do to fix it? Am I stuck with 1.6.7 forever? Mike ___ --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] FW: Polycom 501 issue with latest firmware : sluggish keys
Somebody was helpful enough to give me the very latest release of Polycom's firmware (2.1.0). Unfortunately, I still get that issue. So I'm stuck asking again: Anybody ever got that? Mike _ From: Mike [mailto:[EMAIL PROTECTED] Sent: Wednesday, April 11, 2007 13:37 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Polycom 501 issue with latest firmware : sluggish keys Hi, I've upgraded a few Polycom 501 to SIP 2.0.3b (can't get 2.1.0 because of Polycom's firmware policy, but this is the latest publicly available from Polycom's web site). I've noticed that some keys get sticky though. Soft buttons for example (i.e. end call) need to be pressed 2-3 times for them to react. I've downgraded to 1.6.7, and the problem dissapeared. I can't imagine I'm the only one having that issue, and that issue was also present in 2.0.1 for me. Did anybody else have this problem? What did you do to fix it? Am I stuck with 1.6.7 forever? Mike ___ --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] IMAP Voicemail with MS Exchange
I would love to know if you get this working. We use the SMTP features now, but the ability for a message to be managed from either email client or phone and be changes seen in both is the missing link for us. On 4/11/07, Stephen Bosch [EMAIL PROTECTED] wrote: Anthony Rodgers wrote: Hi there, We're trying to get IMAP voicemail storage working on an MS Exchange server - I would be grateful if anyone who has successfully done this could post the magic soup here, as extensive Google searching has yielded nothing other than tantalizing references to it being done without any specifics. I haven't used IMAP voicemail yet, so you'll have to bear with me here. Have you tried configuring Asterisk to save voicemail messages on the Exchange server using IMAP? What was the result? IMAP support in Exchange, as in Outlook, is rough and rather ugly. For obvious reasons it's never been in MS interest to support it properly, as they want people to use their native Exchange server protocol. There's probably a good reason why you want to do it the way you want to, but I'll ask the question anyway -- what about delivering the message over SMTP? -Stephen- ___ --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 -- Bruce Reeves Nortex Networks ___ --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] Nagios asterisk monitoring
Hi Brandon, On Wednesday 11 April 2007 03:11:38 pm Brandon Kruse wrote: I wrote a very extensive plugin for cacti to monitor asterisk. It uses the manager interface to poll and get statistics for 1.4 and 1.2. Let me know if you interested, ill post it, or email me directly. -bkruse I would be very interested in getting a copy of this for our Cacti install. Thanks, /BAK/ -- Ben Klang Alkaloid Networks 404.475.4850 [EMAIL PROTECTED] http://projects.alkaloid.net ___ --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] Nagios asterisk monitoring
I have a script as well. This actually may be yours Brandon. I found
it through Google. It will just open a telnet session to the manager
interface and count the ZAP and SIP channels. You just have to call
the script through a OID in snmp.conf. Works really well with Cacti.
I will forward it and how it is setup if you like.
http://picasaweb.google.com/jonforrest.beck/AsteriskCLI/photo#5052274842733411794
On 4/11/07, Brandon Kruse [EMAIL PROTECTED] wrote:
I wrote a very extensive plugin for cacti to monitor asterisk.
It uses the manager interface to poll and get statistics for 1.4 and 1.2.
Let me know if you interested, ill post it, or email me directly.
-bkruse
voip crazy wrote:
Dear list,
I am trying to configure the nagios plugin called check_sip. I just
read the README file included with the plugin. I follow the readme
steps to configure the plugin, without success. In the nagios web
interface I can see (No output!) In the status information column. If
I run the chech_sip plugin from a linux console, I get
/usr/local/nagios/libexec# ./check_sip -u sip:[EMAIL PROTECTED]
SIP 200 OK: 0.00 second response time
I do not know why If I run the plugin from the consle it works ok, but
if I run it from Nagios web interface it does not run.
Anyone are using this plugin?
Could you helpme to solve that?
Any clue will be appreciated.
Thanks for your time.
VoipCrazy
Here goes my nagios check_sip plugin configuration.
define command{
command_namecheck_sip
command_line$USER1$/check_sip -u $ARG1$ -H $HOSTADDRESS$ -w 5
}
define service{
use generic-service
host_name -PBX
service_description SIP test
check_command check_sip!sip:[EMAIL PROTECTED]
contact_groups admins
max_check_attempts 4
normal_check_interval 5
retry_check_interval1
notification_interval 240
check_period24x7
notification_period 24x7
notification_optionsc,r
}
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--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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Re: [asterisk-users] Open Source VoIP client (on a webpage)
- Original Message -
From: Tim Panton [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, April 07, 2007 2:44 PM
Subject: Re: [asterisk-users] Open Source VoIP client (on a webpage)
On 6 Apr 2007, at 21:21, Matthew Rubenstein wrote:
On Fri, 2007-04-06 at 12:00 -0700,
[EMAIL PROTECTED] wrote:
Date: Fri, 6 Apr 2007 16:13:29 +0100
From: Tim Panton [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Open Source VoIP client (on a webpage)
To: Jason Wolfe [EMAIL PROTECTED], Asterisk Users Mailing
List - Non-Commercial Discussion asterisk- [EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
On 6 Apr 2007, at 00:59, Jason Wolfe wrote:
I need to decide on the best way to add a voip SIP or IAX client
to
a website. I'm thinking that I'd like it to be inline, like an
aplet, on the page. I've got some asterisk servers running to
connect up to, so the real challenge is finding an easily
integrated open source client.
Any suggestions from those who know?
Our SDK isn't open source, but it is an IAX applet -
javascript/DHTML
friendly and lightweight.
Is that applet available unbundled from the rest of your software and
service package? At a flat (ie not per-instance) price?
Yes, it is available separately and the price is per-server.
Anyone interested should contact me off-list as this is getting
dangerously commercial!
Tim.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
For those interested in commercial WebPhones please see:
http://www.bicomsystems.com/products/C/P/319/382_3543/
Please contact me for more info offline : steve 'at} bicomsystems {dot} com
We also have a very wide range of other products and experience that we will
be able to offer within very reasonable price range.
Steve
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Re: [asterisk-users] FW: Polycom 501 issue with latest firmware : sluggish keys
At 15:09 4/11/2007, Mike wrote: Somebody was helpful enough to give me the very latest release of Polycom's firmware (2.1.0). Unfortunately, I still get that issue. So I'm stuck asking again: Anybody ever got that? Digit Impossible Match in phonexxx.cfg? dialplan dialplan.1.impossibleMatchHandling=2 dialplan.1.removeEndOfDial=1 Mike -- From: Mike [mailto:[EMAIL PROTECTED] Sent: Wednesday, April 11, 2007 13:37 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Polycom 501 issue with latest firmware : sluggish keys Hi, I've upgraded a few Polycom 501 to SIP 2.0.3b (can't get 2.1.0 because of Polycom's firmware policy, but this is the latest publicly available from Polycom's web site). I've noticed that some keys get sticky though. Soft buttons for example (i.e. end call) need to be pressed 2-3 times for them to react. I've downgraded to 1.6.7, and the problem dissapeared. I can't imagine I'm the only one having that issue, and that issue was also present in 2.0.1 for me. Did anybody else have this problem? What did you do to fix it? Am I stuck with 1.6.7 forever? Mike ___ --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] AgentCallBackLogin vs AddQueueMember
I apologize for not responding sooner, I obviously don't read this mailing list regularly. Alan Ferrency wrote: In our investigation of the AddQueueMember vs. AgentCallbackLogin situation, the major loss with using the published AddQueueMember replacement is that it assumes each agent is always using the same phone. I believe I misstated my original meaning, in this quoted statement. Hopefully my explanation below will help clarify our needs. On Wed, 28 Feb 2007, Kevin P. Fleming wrote: This is not true; it is certainly possible to call AddQueueMember() and dynamically construct the channel name that should be added based on the channel that is placing the call to AddQueueMember(). In addition, you can avoid passing _any_ interface name to AddQueueMember() and will use the calling channel name after stripping any suffix present after '-', which in the vast majority of cases will do exactly what you want. This is interesting; I was not aware of the no interface option. However, this is not what we need. This adds a phone channel to the queue, and does not track which person is using that phone. This means that all queue activity is associated with a SIP channel in the logs, which is not acceptable. If two different people log into the same phone at different times, I need the queue log activity to be associated with the person, and not the SIP channel they happen to be using at the time. Later, we need to extract information from the queue log such as the amount of time each person (not the phone) is logged in, the number of calls the person (not the phone) took, how many calls the person abandoned or failed to answer, and so on. We can certainly reimplement the functionality we need, by building a dynamic interface identifier to add to the queue, but there are many aspects of this which are nontrivial. Specifically, to make this effective we would need to maintain a map between a person and the phone that they are using at any particular time. To me, this is the main benefit we get from AgentCallbackLogin. Using this map of people to phones, our dial plan would then need to ensure that: - a person cannot be logged into more than one phone - only one person at a time can be logged into a phone - queue activity logs are associated with a person, not a phone Can the AddQueueMember solution handle the equivalent of autologoff if a queue member fails to answer a queued call in time? To me, saying We removed the AgentCallbackLogin() application because you can reimplement it completely in the dialplan therefore it isn't necessary is just like saying We removed the VoiceMail() application because you can reimplement it in the dialplan. Yes, it's true: these things can be reimplemented in the dial plan. But it's a royal pain in the butt, when what we need already existed. It is also inefficient for every end user who needs the functionality to reimplement it in their own unique way. Thanks for any more help you can give me on this point. I want to believe that the decision to deprecate AgentCallbackLogin makes sense from a standpoint other than decreased code base maintenance cost, but I am still just not seeing it. (I also apologize in advance, if my concerns have been addressed in future asterisk list threads; I'm about to go read those shortly.) Thanks, Alan Ferrency ___ --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] Nagios asterisk monitoring
K, I will be finishing it up this weekend if you want to help beta test. -brandon Ben Klang wrote: Hi Brandon, On Wednesday 11 April 2007 03:11:38 pm Brandon Kruse wrote: I wrote a very extensive plugin for cacti to monitor asterisk. It uses the manager interface to poll and get statistics for 1.4 and 1.2. Let me know if you interested, ill post it, or email me directly. -bkruse I would be very interested in getting a copy of this for our Cacti install. Thanks, /BAK/ ___ --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 - Static IP
Hi Steve - Is there a way to config a static ip address on a Polycom phone remotely ie. From a config file or a web browser? If you have a good DHCP server, you can use it to assign a static address to the phone's MAC. What DHCP server are you using? - Noah ___ --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] Nagios asterisk monitoring
thats the same idea forrest, correct.
but if the need is great enough, I want to rewrite it with
more features and default templates and other things like that.
If anyone is interested in me rewriting it, send your feature
request and a vote to [EMAIL PROTECTED]
We will see how it turns out ;]
-brandon
Forrest Beck wrote:
I have a script as well. This actually may be yours Brandon. I found
it through Google. It will just open a telnet session to the manager
interface and count the ZAP and SIP channels. You just have to call
the script through a OID in snmp.conf. Works really well with Cacti.
I will forward it and how it is setup if you like.
http://picasaweb.google.com/jonforrest.beck/AsteriskCLI/photo#5052274842733411794
On 4/11/07, Brandon Kruse [EMAIL PROTECTED] wrote:
I wrote a very extensive plugin for cacti to monitor asterisk.
It uses the manager interface to poll and get statistics for 1.4 and
1.2.
Let me know if you interested, ill post it, or email me directly.
-bkruse
voip crazy wrote:
Dear list,
I am trying to configure the nagios plugin called check_sip. I just
read the README file included with the plugin. I follow the readme
steps to configure the plugin, without success. In the nagios web
interface I can see (No output!) In the status information column. If
I run the chech_sip plugin from a linux console, I get
/usr/local/nagios/libexec# ./check_sip -u sip:[EMAIL PROTECTED]
SIP 200 OK: 0.00 second response time
I do not know why If I run the plugin from the consle it works ok, but
if I run it from Nagios web interface it does not run.
Anyone are using this plugin?
Could you helpme to solve that?
Any clue will be appreciated.
Thanks for your time.
VoipCrazy
Here goes my nagios check_sip plugin configuration.
define command{
command_namecheck_sip
command_line$USER1$/check_sip -u $ARG1$ -H $HOSTADDRESS$
-w 5
}
define service{
use generic-service
host_name -PBX
service_description SIP test
check_command check_sip!sip:[EMAIL PROTECTED]
contact_groups admins
max_check_attempts 4
normal_check_interval 5
retry_check_interval1
notification_interval 240
check_period24x7
notification_period 24x7
notification_optionsc,r
}
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Re: [asterisk-users] FW: Polycom 501 issue with latest firmware : sluggish keys
Hi Mike - Somebody was helpful enough to give me the very latest release of Polycom's firmware (2.1.0). Unfortunately, I still get that issue. So I'm stuck asking again: Anybody ever got that? I've got quite a few Polycoms of various models running in a number of asterisk installs. Some of them are on 1.6.7, but most are on 2.0.3 or 2.1.0. I haven't seen this one at all. I would definitely call your reseller to have them bring it up with Polycom. If your reseller won't take the time, you may be able to find others that will - if you buy a phone from them ;-). www.voipsupply.com comes to mind, but I'm sure there are other vendors who will go to bat for you. - Noah ___ --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] Nagios asterisk monitoring
On Wednesday 11 April 2007 03:11:38 pm Brandon Kruse wrote: I wrote a very extensive plugin for cacti to monitor asterisk. It uses the manager interface to poll and get statistics for 1.4 and 1.2. Let me know if you interested, ill post it, or email me directly. -bkruse I did not appreciate how cool this was until I researched RRDTool and Cacti! I am definitely interested in this as well. I have a feeling that many in the * community will want to learn more about this. Thanks, MC ___ --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] AgentCallBackLogin vs AddQueueMember
Alan Ferrency wrote: However, this is not what we need. This adds a phone channel to the queue, and does not track which person is using that phone. This means that all queue activity is associated with a SIP channel in the logs, which is not acceptable. Right. This is why we added the 'membername' argument to the AddQueueMember application, so that queue logs can reflect a logical name for the queue member, regardless of what channel/interface they logged in from. Using this map of people to phones, our dial plan would then need to ensure that: - a person cannot be logged into more than one phone - only one person at a time can be logged into a phone - queue activity logs are associated with a person, not a phone For points #1 and #2, you are correct that this logic will have to be built. Point #3 is already taken care of by the addition of the 'membername' as I commented on above. However, I personally see this as a huge benefit; I much prefer Asterisk to provide mechanisms for users to do things, but not the policy on how they are to be used. When chan_agent is in use, you don't get to decide what to do if a second user tries to log in from the same channel, that has been decided for you. If instead you write that logic in the dialplan (or start from an example you find in the docs, on the wiki, etc.) you can completely control how the system behaves. Can the AddQueueMember solution handle the equivalent of autologoff if a queue member fails to answer a queued call in time? Absolutely; the example in doc/queues-with-callback-members.txt shows how to do it. To me, saying We removed the AgentCallbackLogin() application because you can reimplement it completely in the dialplan therefore it isn't necessary is just like saying We removed the VoiceMail() application because you can reimplement it in the dialplan. If that was true, we would have already done it. In fact there is an effort under way to do exactly that, and for the reason I outlined above: today, if you want the voicemail system to behave slightly (or significantly) differently, you must modify the C code to do it. This is in spite of the fact that the voicemail system is just a fancy IVR, and we already have plenty of ways to build IVRs in Asterisk. Olle Johansson's 'minivm' branch is an attempt to work towards fixing this, so that the important voicemail-specific parts of app_voicemail will be available as individual dialplan applications, but the 'personality' of the voicemail system will be defined by the IVR the administrator chooses to wrap around them. Yes, it's true: these things can be reimplemented in the dial plan. But it's a royal pain in the butt, when what we need already existed. It is also inefficient for every end user who needs the functionality to reimplement it in their own unique way. You will have AgentCallbackLogin at your disposal until Asterisk 1.4 no longer suits your needs, which could be years from now. There is no reason for you to do _anything_ today, other than to start thinking about how you want to do it in the future when you decide to upgrade to Asterisk 1.6 and have to replace it. If there is no simple replacement available to you at that time (which would be highly surprising considering that it already exists today) then I can see your point, but acting today like the functionality has been removed and that you are being forced to rearchitect your system seems a little bit extreme (in my opinion, of course). ___ --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] Open Source VoIP client (on a webpage)
-Original Message-
On 6 Apr 2007, at 21:21, Matthew Rubenstein wrote:
On Fri, 2007-04-06 at 12:00 -0700,
[EMAIL PROTECTED] wrote:
Date: Fri, 6 Apr 2007 16:13:29 +0100
On 6 Apr 2007, at 00:59, Jason Wolfe wrote:
I need to decide on the best way to add a voip SIP or IAX client
to
a website. I'm thinking that I'd like it to be inline, like an
aplet, on the page. I've got some asterisk servers running to
connect up to, so the real challenge is finding an easily
integrated open source client.
Any suggestions from those who know?
Our SDK isn't open source, but it is an IAX applet -
javascript/DHTML
friendly and lightweight.
Is that applet available unbundled from the rest of your software
and
service package? At a flat (ie not per-instance) price?
Yes, it is available separately and the price is per-server.
Anyone interested should contact me off-list as this is getting
dangerously commercial!
Tim.
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
For those interested in commercial WebPhones please see:
http://www.bicomsystems.com/products/C/P/319/382_3543/
Please contact me for more info offline : steve 'at} bicomsystems
{dot} com
We also have a very wide range of other products and experience that
we will
be able to offer within very reasonable price range.
Steve
___
Just to be clear, the Jiax client that Bicom is offering is different to
the Mexuar Corraleta SDK.
Not that I am saying one or the other is better :-), just making sure
people know that there is a difference.
Cheers,
Dean Collins
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
http://click.mexuar.com/webuser/click/7/userurl/Cognation
http://click.mexuar.com/webuser/nojs/7/userurl/Cognation
www.Mexuar.com http://www.mexuar.com/
Want to voice enable your website?
Use Corraleta to reach your customers in 10 seconds or less.
image001.gif
Description: image001.gif
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[asterisk-users] Which SIP phones to buy?
I need to buy some new phones for our own offices. I've used only Polycom phones until now, but I'd like to broaden my experience. I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I don't want to broaden my knowledge. Advice, anyone? -Stephen- ___ --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] help with Sipura SPA 3000
On 10 de abr de 2007, at 23:05, James Harper wrote:
2 - How can I gain full control to the FXS? I mean, a simple * dialed
is
not sent for asterisk (the server) interpretation, probably because
it's
used by Sipura's suplementary services, I don't know. Also, is it
possible
to get a dial tone from ASterisk, instead of Sipura's? My goal with
this
is to provide users with direct access to the PSTN line pressing 0,
instead of collecting calls and making the call themselves, or at
least
making ignorepat to work!
A dialplan of '(S0:s)' will get your phone to jump straight into the
's' extension in asterisk as soon as someone picks it up. From
there you
can do something like:
It worked perfectly! Thanks!
[sip_ata_incoming]
exten = s,1,Answer
exten = s,n,DISA(no-password|sip_extension_in)
so Asterisk will give you dialtone and do the dialplan stuff for you.
From the 'sip_extension_in' context you can make a single '0' or '*'
call the PSTN line.
On the sip_extension_in, I entered the following
exten = 0,1,NoOp(outgoing call via POTS line to No. ${EXTEN:1})
exten = 0,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],120)
exten = 0,3,Congestion()
exten = 0,4,Hangup
However, when I press the 0, it does gives me a dialtone, but it
doesn't seem to be delivering the tones imediately. I even suspect it
isn't my PSTN tone after the 0. Is there something else?
Cheers,
Francis
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Re: [asterisk-users] Nagios asterisk monitoring
Yes, I have actually written a resource module for asterisk and the gui to use rrdtool to make REAL pretty gradient shaded graphs based on asterisk data. So, if you want the cacti script, email me([EMAIL PROTECTED]) to get me motivated to rewrite it and make it awesome, and encouragement would be great. But, with a pbx not a pretty graph maker, I am now working on clientside graphing using svg(z) and doing httprequests to get manager information. Let me know if you are interested in that also, I didnt realize how much of a community was out there for monitoring :] -brandon Michael Collins wrote: On Wednesday 11 April 2007 03:11:38 pm Brandon Kruse wrote: I wrote a very extensive plugin for cacti to monitor asterisk. It uses the manager interface to poll and get statistics for 1.4 and 1.2. Let me know if you interested, ill post it, or email me directly. -bkruse I did not appreciate how cool this was until I researched RRDTool and Cacti! I am definitely interested in this as well. I have a feeling that many in the * community will want to learn more about this. Thanks, MC ___ --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] Polycom 501 issue with latest firmware : sluggish keys
Jim King wrote: I've seen an issue like this from time to time on 601s, even with the latest firmware. Not just the softkeys, but also the dial keys. The phones seem to run slow sometimes, failing to respond to a key press right away but getting to it eventually. It usually clears up after a few seconds. Also, I've noticed that the 601s sometimes ignore key presses altogether, just as you describe. I have not yet found a solution for this problem... Try setting this in sip.cfg: dialplan.impossibleMatchHandling=1 I suspect it is either 0 or 2 now. ___ --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] Nagios asterisk monitoring
Allow me to register my interest in any and all things that tie Asterisk information to Cacti. We use that here, and it's been on my to-do list for a lgg time. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brandon Kruse Sent: Wednesday, April 11, 2007 6:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Nagios asterisk monitoring Yes, I have actually written a resource module for asterisk and the gui to use rrdtool to make REAL pretty gradient shaded graphs based on asterisk data. So, if you want the cacti script, email me([EMAIL PROTECTED]) to get me motivated to rewrite it and make it awesome, and encouragement would be great. But, with a pbx not a pretty graph maker, I am now working on clientside graphing using svg(z) and doing httprequests to get manager information. Let me know if you are interested in that also, I didnt realize how much of a community was out there for monitoring :] -brandon The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] Nagios asterisk monitoring
Let me also add my interest, we've got a site using Nagios and haven't had time to work anything out yet related to Asterisk. Cheers, Joel. Joel Hill Support Engineer Asterisk IT On Wed, 2007-04-11 at 18:42 -0400, Watkins, Bradley wrote: Allow me to register my interest in any and all things that tie Asterisk information to Cacti. We use that here, and it's been on my to-do list for a lgg time. - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brandon Kruse Sent: Wednesday, April 11, 2007 6:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Nagios asterisk monitoring Yes, I have actually written a resource module for asterisk and the gui to use rrdtool to make REAL pretty gradient shaded graphs based on asterisk data. So, if you want the cacti script, email me([EMAIL PROTECTED]) to get me motivated to rewrite it and make it awesome, and encouragement would be great. But, with a pbx not a pretty graph maker, I am now working on clientside graphing using svg(z) and doing httprequests to get manager information. Let me know if you are interested in that also, I didnt realize how much of a community was out there for monitoring :] -brandon The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] ZAP does not disconnect
I have a ZAPTEL interface card with 4 channel. If I call out through the zap channel to my mobile, the mobile starts ringing, but If I disconnect the internal phone that is my SIP client the mobile does not stop ringing. Anyone any suggestion of what am I doing wrong. Regards, Sanjay Rajdev Phone : +1 (877) 342 2329 x 1702 Fax : +1 (815) 261 5907 http://www.featherstoneinformatics.com Communications from Featherstone Informatics Group (FIG) may transmit information that is confidential and privileged information of Featherstone Informatics Group (FIG). Unless you are the intended addressee, you may not use, copy or disclose to anyone this communication or any information transmitted by this communication. If you have received such communication in error, please advise the sender by e-mail and/or telephone and destroy this communication immediately. This communication and any information transmitted by this communication may also be considered protected health information as defined under the Health Insurance Portability and Accountability Act and its related regulations (a.k.a., HIPAA) or any other similar state law. Please exercise due care and ensure that you comply with its contractual and legal obligations. ___ --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] Which SIP phones to buy?
Stephen Bosch wrote: I need to buy some new phones for our own offices. I've used only Polycom phones until now, but I'd like to broaden my experience. I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I don't want to broaden my knowledge... ...because I like to stay dumb. Of course, that's not what I meant :) -Stephen- ___ --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] What is your Backup Strategy?
I was just curious to what your redundancy solution is. I have considered many options, so I thought I would share and get an idea for what others are doing. My setup is two different locations with a 10MB WLAN fiber link between the two. Each location has it's own PRI as well. I have considered and tested many options this last year or so. 1) Using hearbeat and drbd to monitor the servers. When the primary fails the backup will assign itself the virtual ip used between the two, and then mount the drbd disk which has the asterisk configs and voicemail. The biggest con to this is hearbeat just monitors a ping response either over IP or a COM port. So if the asterisk service dies, heartbeat will not fail over. Although I think there are work arounds for this. The newest version is suppose to have support for monitoring a TCP port as well 2) Have two servers with the same dialplan. One in each location. Each server has it's own TDM cards installed. Phones on Site A will register with the server on Site A, and phones on Site B will register with the server on Site B. Then using Polycom phones, they will failover to using the server not on their site, if their primary isn't available. I have setup scripts to copy the dialplan from one server to the other then reload asterisk nightly. The biggest Con to this is I have to be sure my dialplans don't get different. The user's voicemail wouldn't be available until their primary server is back up, but that's OK. 3) Having a main asterisk server and a smaller VoIP gateway at each site. The gateway is a small 14inch deep rack server with a P4 and 1Gig RAM running asterisk. It will host the TDM cards, and just handle traffic to/from the PRI. The main asterisk server will just see it as a SIP trunk. The failover here is that the polycom phones will register with the gateway if the primary server isn't available. They won't have all the features and voicemail, but at least they can dial out and get 911 if needed. What do you think? Do you have a better solution? Thanks!! -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --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] help with Sipura SPA 3000
A dialplan of '(S0:s)' will get your phone to jump straight into
the
's' extension in asterisk as soon as someone picks it up. From
there you
can do something like:
It worked perfectly! Thanks!
Just remember that having Asterisk supply the dialtone does add (a
slight) additional load, rather than it just routing calls between
endpoints. Not an issue with one or two ATA's though.
[sip_ata_incoming]
exten = s,1,Answer
exten = s,n,DISA(no-password|sip_extension_in)
so Asterisk will give you dialtone and do the dialplan stuff for
you.
From the 'sip_extension_in' context you can make a single '0' or
'*'
call the PSTN line.
On the sip_extension_in, I entered the following
exten = 0,1,NoOp(outgoing call via POTS line to No. ${EXTEN:1})
exten = 0,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],120)
exten = 0,3,Congestion()
exten = 0,4,Hangup
However, when I press the 0, it does gives me a dialtone, but it
doesn't seem to be delivering the tones imediately. I even suspect it
isn't my PSTN tone after the 0. Is there something else?
A few things to check:
. ${EXTEN:1} will be empty because the extension can only be '0'. Change
it to 'SIP/LinkSysOut' instead
. I'm not sure but I think that the SPA3000 can either present a 'false'
dialtone to the SIP call on the PSTN line, take the digits, then send
them to the PSTN then connect the SIP call to it, or it can give the
real PSTN dialtone and connect the call immediately. I think the latter
is what you want but I can't remember the name of the setting. Maybe
'one stage dialling'?
. Related to the above, I think you might need to set the dialplan on
the VoIP to PSTN settings to 'none'.
HTH
James
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Re: [asterisk-users] What is your Backup Strategy?
On 4/11/07, Forrest Beck [EMAIL PROTECTED] wrote: 2) Have two servers with the same dialplan. One in each location. Each server has it's own TDM cards installed. Phones on Site A will register with the server on Site A, and phones on Site B will register with the server on Site B. Then using Polycom phones, they will failover to using the server not on their site, if their primary isn't available. I have setup scripts to copy the dialplan from one server to the other then reload asterisk nightly. The biggest Con to this is I have to be sure my dialplans don't get different. The user's voicemail wouldn't be available until their primary server is back up, but that's OK. Now you said you had two machines with the same dialplans. What happens when you go into fail over an someone leaves a voice message and it gets stuck on the other server? I think the key here is to treat functions as a cluster. IVRs, voicemail, phone calls, etc you need to have a redundant solution for each, not just a spare or redundant asterisk server. Then again you could be working on a small scale project where what I describe its not really important. ___ --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] What is your Backup Strategy?
On Wed, 11 Apr 2007, Forrest Beck said something to this effect: 1) Using hearbeat and drbd to monitor the servers. When the primary fails the backup will assign itself the virtual ip used between the two, and then mount the drbd disk which has the asterisk configs and voicemail. The biggest con to this is hearbeat just monitors a ping response either over IP or a COM port. So if the asterisk service dies, heartbeat will not fail over. Although I think there are work arounds for this. The newest version is suppose to have support for monitoring a TCP port as well This seems like a good approach, if you've got any stability and/or filesystem-related quirks ironed out -- I've heard of some. I don't know much about heartbeat, but I don't imagine it'd be hard to hack in a SIP polling event either internally or externally. You could use SIP Swiss Army Knife (sipsak) or some other SIP testing tool to send a periodic OPTIONS ping to the SIP service and trigger a protection switch to the secondary server if it's down. Even if you can't hack this into the heartbeat setup itself (can't it use external scripts for monitoring?), you can certainly do something like run it on the primary server and if the SIP service dies, enact a firewall rule that drops ICMP responses and thus artificially trigger a failure. -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --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] What is your Backup Strategy?
On Wed, 11 Apr 2007, Andrew Joakimsen said something to this effect: Now you said you had two machines with the same dialplans. What happens when you go into fail over an someone leaves a voice message and it gets stuck on the other server? I would agree strongly. This is why I favour either a Linux high-availability cluster (HA heartbeat) setup, or some kind of intermediate storage layer for more sophisticated messaging systems that is replicated and which all the nodes can access uniformly. -- Alex Balashov [EMAIL PROTECTED] ___ --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] What is your Backup Strategy?
We are looking at about 200 total phones with low usage. Probably only 20 or so calls at once. On 4/11/07, Andrew Joakimsen [EMAIL PROTECTED] wrote: On 4/11/07, Forrest Beck [EMAIL PROTECTED] wrote: 2) Have two servers with the same dialplan. One in each location. Each server has it's own TDM cards installed. Phones on Site A will register with the server on Site A, and phones on Site B will register with the server on Site B. Then using Polycom phones, they will failover to using the server not on their site, if their primary isn't available. I have setup scripts to copy the dialplan from one server to the other then reload asterisk nightly. The biggest Con to this is I have to be sure my dialplans don't get different. The user's voicemail wouldn't be available until their primary server is back up, but that's OK. Now you said you had two machines with the same dialplans. What happens when you go into fail over an someone leaves a voice message and it gets stuck on the other server? I think the key here is to treat functions as a cluster. IVRs, voicemail, phone calls, etc you need to have a redundant solution for each, not just a spare or redundant asterisk server. Then again you could be working on a small scale project where what I describe its not really important. ___ --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 -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --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] Which SIP phones to buy?
Hi Stephen, I'm using Grandstream and I think is a nice phone, but its the only one that I've tried. I bought it to learn voip/asterisk. Just my 2 cents. Good luck. Ronaldo. On 4/11/07, Stephen Bosch [EMAIL PROTECTED] wrote: Stephen Bosch wrote: I need to buy some new phones for our own offices. I've used only Polycom phones until now, but I'd like to broaden my experience. I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I don't want to broaden my knowledge... ...because I like to stay dumb. Of course, that's not what I meant :) -Stephen- ___ --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] adding chan_celliax
Tzafrir Cohen wrote: On Mon, Apr 09, 2007 at 04:41:42PM -0500, Patricio Valarezo Lozano wrote: Hi, thanks a lot for your directions, I've downloaded the svn from celliax, but i wasn't aware than these sources are the full asterisk sources including asterisk and chan_celliax. So, i thinks than it's a good idea to compile the sources, build a deb and forget about the debian official deb. Is that right or there is a more debian way to add this channel to the debian release. Debian already has several packages of out-of-tree asterik modules, that are built vs. asterisk-devel: * asterisk-addons * The spandsp modules (fax, dtmftotext) * chan_capi-cm This is how I built my deb. mmm... may you post the general directions to add the sources from celliax to the debian asterisk-devel? i'll get a new asterisk .deb or just get the module in .deb format to add to the regular asterisk .deb?? sorry if my questions are a little basic, i don't have experience modding sources from debian devel packages. thank you -- patoVala Linux User#280504 Hablando en http://www.elprimoalcahuete.com ... at least I thought I was dancing, 'til somebody stepped on my hand. -- J. B. White ___ --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
