[Asterisk-Users] Re: Get no busy signal on my analog line
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Does this belong to my dialplan or my sip registration settings? To your SIP registration settings. You should limit that user/peer/friend to only one line. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk management interface
[EMAIL PROTECTED] wrote: Looks very nice.. Is it GPL, GNU? PBXware interface is not GPL/GNU currently. Some time in the future we may release is it under GPL/GNU license :)... ___ --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:Toshiba DK424 / Asterisk / DTMF problems
Anthony, I will suggest you to use E1, you got 30 channels to communicate. I did the integration with Toshiba CTX using E1, and no problem at all. Asterisk as Pri_net Toshiba as PRI_cpe /etc/zaptel.conf span=1,1,0,ccs,hdb3 ;this is depanding on your pbx settingyou got to test it out, you could place crc4 or omit... bchan=1-15 dchan=16 bchan=17-31 loadzone=sg defaultzone=sg context=toshiba-intercom group=2 usecallerid=yes hidecallerid=no transfer=yes cancallforward=yes echocancel=yes echotraining=yes busydetect=yes busycount=2 immediate=no ;context=from-zap-trunk-1 switchtype=euroisdn overlapdial=yes signalling=pri_net pridialplan=local priindication=outofband channel=1-15 channel=17-31 Message: 3 Date: Thu, 2 Mar 2006 11:58:22 -0500 From: Anthony Cennami [EMAIL PROTECTED] Subject: [Asterisk-Users] Toshiba DK424 / Asterisk / DTMF problems To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I have a Toshiba DK424 connected via T1 EM to a TE110P card. Intermittently when a user dials a number I am getting 'getdtmf' errors on the Ast server and the calls do not go through. If they dial the number once or twice more, it works fine and I receive no DTMF problems. On another note, end users are complaining about intermittent disconnects. T1 is ESF/B8ZS - 24 chan. Other than those two problems the voice quality appears OK and I haven't really seen too many other problems. If there's anybody here running a similar config can you let me know if you've encountered this and what solutions you've devised. zapata.conf [trunkgroups] [channels] language=en group=1 context=from-pbx signalling=em_w relaxdtmf=yes usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=no callwaitingcallerid=no threewaycalling=no transfer=no canpark=no cancallforward=no callreturn=no echocancel=no echocancelwhenbridged=no busydetect=yes rxgain=5.0 txgain=5.0 callgroup=1 pickupgroup=1 immediate=no channel = 1-24 [zaptel.conf] span=1,1,0,esf,b8zs em=1-24 -- Anthony D Cennami -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060302/16e364 38/attachment-0001.htm -- ___ --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] Status of another channel from AGI
I have an AGI program with an array containing a set of ${UNIQUEID}
variables for channels that may be active on the system. I need a way
for the program to tell if they are or not.
It's certainly possible using the manager interface, or appropriate
asterisk -rx commands, but I'd prefer to do it directly from AGI for
performance, security, and ease of configuration.
Does anyone know a neat way of doing this?
Does anyone know a neater way using console commands to get the
uniqueids on the system than show channels concise, then for each row
returned a show channel channel and parsing for uniqueid?
All systems this will run on are Asterisk 1.2.4 or higher.
--
Alistair Cunningham,
Integrics Ltd,
+44 20 799 39 799
sip:[EMAIL PROTECTED]
http://integrics.com/
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Re: [Asterisk-Users] Re: Asterisk at large
On Thu, Mar 02, 2006 at 04:14:34PM -0700, Douglas Garstang wrote: The best way to achieve maximum manageability is to design a MySQL database and develop AGI scripts (in your language of choice) that work to that design. I've found that it has been far easier to develop complex routing logic in python than it is in the horrible assembler-like Asterisk extensions.conf language. In a perfect world, there would be _ONE_ line in your extensions.conf, and it would be: exten = _.,1,AGI(routing_script) agreed. I'm building a system for a few hundred users and this is the method I'm using basically. You'd also replicate your databases and put logic into your application such that upon failure to connect to the primary database, the application can seamlessly start performing reads from the secondary, replicated server. Doug. -Original Message- From: Cosmin Prund [mailto:[EMAIL PROTECTED] Sent: Thursday, March 02, 2006 1:26 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Re: Asterisk at large I'm no Asterisk genie but if you're running on one server you're probably not dealing with lots of users (for what I'm trying to say 100 users are not a lot of users). Factoring in the VERY simple format of both sip.conf and extensions.conf, isn't it possible to create an php page that would generate those two files from the database? You'll next need to run a basic script that would call the php's + asterisk -rx reload and you'd be done! If you're trying to skip the reload step (ie: make the changes available immediately / transparently) I don't think it can be done, and this has nothing to do with Asterisk and a lot more to do with databases. Asterisk is something outside the database, using the database as nothing more but a source for data. Asterisk will not know the data in the database has changed, it needs to be told! On the other hand I am a newbie to Asterisk and I don't really like/know mySql so I might be very wrong and far from the truth. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C F Sent: Thursday, March 02, 2006 9:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Asterisk at large Douglas, a lot easier? If it's like you say with multiple servers. But the OP did not indicate this in his/her question, in fact s/he sounded clueless. Also, what is the purpose of NOT having *any* configs from /etc/asterisk/ On 3/2/06, Douglas Garstang [EMAIL PROTECTED] wrote: Yikes. Managability! It's a lot easier to manage multiple Asterisk systems configuration from a single MySQL database then it is to manage .conf files on several redundant Asterisk boxes. I can't believe you asked that question. I'll apologise in advance because I must be missing part of this thread. -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Thursday, March 02, 2006 10:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Asterisk at large Can you explain why you would want asterisk only thru realtime? and not thru the /etc/asterisk/ ? The wiki is located at: http://www.voip-info.org/ the archives for this list is located at: http://lists.digium.com/ The asterisk irc channel is at: irc://irc.freenode.net/#asterisk Google is located at: http://www.google.com/ The asterisk docs project is located at: http://www.asteriskdocs.org/ On 3/2/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi Group, Please read my previous message below, I want to configure Asterisk with Mysql and make Asterisk dynamic so that Asterisk will read everything from Mysql and we can make changes to mysql data directly. Please tell how can we do this and point me to related documentation. Thanks for your help and time, Manoj. Quoting [EMAIL PROTECTED]: Hi Group, I was able to install Asterisk and its addons successfully. Now I want to eliminate sip.conf and extensions.conf and use everything from Mysql DB, Is this possible? I have seen this page http://www.voip- info.org/wiki/index.php?page=Asterisk%20extensions%20from%20mysql and learnt that we still get the data from Mysql DB and write it as sub file to actual sip or extensions.conf before starting Asterisk. Can we eliminate config files completely? If it is possible then please point me to the links explaing how can we do this? I also found very less information on using Asterisk with Mysql, if there are any articles discussing this please send me those links. Thanks for your help all the time, Manoj.
RE: [Asterisk-Users] Polling Asterisk for Life
So, simply respawning asterisk, or checking to see if it's running isn't good enough, because asterisk is indeed running. We need to access asterisk and issue a command, and see if asterisk responds appropriately. If not, we can assume it has died, and we can kill it off (killall -9 asterisk) and then start it back up again (or reboot the whole server if necessary). The _only_ way to reliably (well, in as much as that is possible) to test if your Asterisk is working, is to build a monitoring system that does more or less the same as a typical user would do. We have a system with two modems connected to ATA's and they dial each other via multiple routes so we test all of the major scenarios. We only test if calls are routed through, not if the call itself establishes (media running) to prevent major costs from such a system. I works reasonably well, it seems to detect 99% of the major problems. -- Andreas Sikkema BBned NV Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --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] Problem with NAT!!!
Hi all I'm a newbie in asterisk.I install asterisk server successfully. I configure this server to traverse NAT. Using Xlite clients, i make a call between 2 local networks through Internet.Asterisk server is installed on a host with public IP. client A (in the LAn A) and client B (in the LAN B) are registered. When i make a call from the LAN A to the LAn B, everything goes well.But, when i try to make a call from the Lan B to the Lan A, the xlite client B, displays: connecting and after a time, it display timeout 408:call failed. CAn anybody explain me what happen? Bests regards ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.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
[Asterisk-Users] Re: Native music on hold - Error
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... what are the file permissions/ownership and are they readable by the asterisk process ? The problem was that wav files where in stereo mode. I have encode them and now it works fine. -- Tomislav Parcina [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] is there a variable for the calling IP ?
I know there's a variable for the IP of a SIP channel, but I can't find if such a variable is avaliable for a generic voip cahnnel, or at least h323 channels (ooh323) ___ --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] Part-Time work available
Hi, I'm looking for someone to do time-to-time mantainence on some of our machines going up in New York. The person *MUST* be stationed in New York. Areas of expertise required: - Proficiency in Linux: Slackware, Fedora - Proficiency on Cisco Routers If anybody is interested, please contact me off-list. Regards, Sahil Gupta VoiceValley ___ --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: a2billing without IVR
ram wrote: Hi how about when trying to call SIP extention to SIP extension Local cal even though its going to out route when i enable SIP_IAX=YES then its IVR in place ask 9 to dial SIP/IAX, if not its dial to international call How can i avoide this check if the user belong to local, dial directly if its international call, go to out route any idea, how can i achieve this Sure, just do this in your dialplan. In the a2billing context, list your local extentions first, then the a2billing ones. If the caller is calling a local extension this will get picked up instead of a2billing. I do it like this: 1. Create the a2billing context: [a2billing] exten = s,1,Answer exten = s,2,Wait,2 exten = s,3,DeadAGI(a2billing.php|1) exten = s,4,Wait,2 exten = s,5,Hangup 2. Create your local extensions context: [local] 8001 = 8002 = etc. 3. Create the actual context that you send users into, which does an include of the [local] and [a2billing] contexts: [a2users] include = local include = a2billing Now, local calls will not go anywhere near a2billing. Hope this helps. -Barry ram On 2/26/06, *Guillermo Salas M* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Fri, 2006-02-24 at 10:58 +, Barry Flanagan wrote: Asterisk Sales wrote: mailto: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Hello list, Is there any way to use a2billing without the IVR for the sip/iax users. (authentication is done by the user id and pass as user registers with asterisk). I want to dial the destination number to the asterisk. for example: user dials, exten =_011.,1,DeadAGI(a2billing) system will connect the destination and bill them. but right now we need to enter the destination followed by the IVR prompts which i dont want. Thanks in advanved if anybody can help me. Yes, this is all configurable from /etc/asterisk/a2billing.conf If you set use_dnid=YES then a2billing will pick up the destination from the number the user dialled. Set the following to turn off the IVR stuff: ; Play the balance to the user after the authentication (values : yes - no) say_balance_after_auth=NO ; Play the balance to the user after the call (values : yes - no) say_balance_after_call=NO ; Play the time the user can call (values : yes - no) say_timetocall=NO Hope this helps. Thank you, is working for me right now :) -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --Bandwidth and Colocation provided by Easynews.com http://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 -- -Barry Flanagan ___ --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 only for logging users
hi all i have a asterisk configured and working perfectly. but i have a problem. if i download a softphone for example sjphone and digit for example [EMAIL PROTECTED] i receive this call. is possible to block this? i only want to received calls for login 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] login/logout agents in a specific queue
hi if i have an agents that figure as a member in more than one queue, how can i login / logout him in a specific queue an not in all queues? ___ --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] Implementing MOH while trunks gets connected...
Hi, I just want to implement Music on Hold while my * tried to get the connection on the trunks. Any clues would be welcome... It needed for me as when failover trunk sometimes take effect, it takes sometime for the connection and unknowingly we disconnect the call. MOH is just to avoid this. Thanks in advance... Dan On 03/03/06, nik600 [EMAIL PROTECTED] wrote: hi if i have an agents that figure as a member in more than one queue, how can i login / logout him in a specific queue an not in all queues? ___ --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] Polling Asterisk for Life
On 3/3/06, Martin Joseph [EMAIL PROTECTED] wrote: On Mar 2, 2006, at 9:46 AM, Matt wrote: Doesn't it seem absurd to go through all these gyrations, rather then troubleshooting and fixing the problem? I know you have already tried without success, but this seems absurd to me. I am discomforted by the number of people saying they are rebooting nightly and have cron jobs to restart. Yes it does.. but unfortunately until someone or myself can figure out what is causing the lockups it is the only solution. (P.S. I prefer reloads to reboots.. no need to reboot the system). Anyway, to answer the other e-mailer... I am running a single port TE110P ___ --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: Get no busy signal on my analog line
On 3/3/06, Tomislav Parčina [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Does this belong to my dialplan or my sip registration settings? To your SIP registration settings. You should limit that user/peer/friend to only one line. How is the best way to limit it? []'s -- Tomislav Parcina tparcina#lama.hr ___ --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 -- Bruno de Assumpção Loureiro msn: [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] web meetme instructions
This has to be the worst documentation I have ever come acrossed. I have found two or three docs on how to install it, but they are all so different and make huge assumption about what packages you have installed and locations of files. Has anyone seen something better, I want to get this working it is quite a cool app. Jordan Novak Communications Technician Logistics Health Inc. 1319 Saint Andrews Street La Crosse WI 54603 ___ --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] web meetme instructions
First things first... use the latest version... (that I know of) http://www.fitawi.com/Asterisk/ second... which part are you having problems with? The web piece? or the app_cbmysql? For the app_cbmysql, I have found that the easiest way to work with it is to incorperate it into asterisk-addons... copy app_cbmysql.c into /usr/src/asterisk-addons-1.2.x add MODS+=app_cbmysql.so after MODS=format_mp3... make make clean The web portion is pretty straight forward... just make sure you have register_global=On in php.ini. That being said... I am personally not a big fan of the web portion of the interface ;) I have written my own that allows users to create there own based on their voicemail login. Sean On Fri, 2006-03-03 at 07:31 -0600, Jordan Novak wrote: ___ --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 Fax Question
Hi All, I want to configure fax with Asterisk and I found that we can do this reliably using G711 codec only. Currently my provider is supporting G729 and G711. During the call initiation the call starts with G729 (1'st priority) and somehow if the receiver is unable to receive call then we are providing the Caller to send a fax, but at that point they are using G729 codec. At this point how can the user send the fax using G711? We want to use G711 only as Fax is delivered reliably using it. Please tell me whether this is possible and if possible how can we achieve it after the call is accepted with G729? Thanks for your help and time, Manoj. ___ --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 Realtime Configs Samples with MySQL
I haven't tried sip yet... been finishing voicemail, but the principal is the same. res_mysql.conf [general] dbhost = localhost dbname = asterisk dbuser = someuser dbpass = somepass dbport = 3306 dbsock = /var/run/mysqld/mysqld.sock extconfig.conf voicemail = mysql,asterisk,voicemail ; i would assume that sip would be sippeers = mysql,asterisk,sip sipusers = mysql,asterisk,sip Table CREATE TABLE `sip` ( `id` int(11) NOT NULL auto_increment, `name` varchar(80) NOT NULL default '', `accountcode` varchar(20) default NULL, `amaflags` varchar(13) default NULL, `callgroup` varchar(10) default NULL, `callerid` varchar(80) default NULL, `canreinvite` char(3) default 'yes', `context` varchar(80) default NULL, `defaultip` varchar(15) default NULL, `dtmfmode` varchar(7) default NULL, `fromuser` varchar(80) default NULL, `fromdomain` varchar(80) default NULL, `fullcontact` varchar(80) default NULL, `host` varchar(31) NOT NULL default '', `insecure` varchar(4) default NULL, `language` char(2) default NULL, `mailbox` varchar(50) default NULL, `md5secret` varchar(80) default NULL, `nat` varchar(5) NOT NULL default 'no', `deny` varchar(95) default NULL, `permit` varchar(95) default NULL, `mask` varchar(95) default NULL, `pickupgroup` varchar(10) default NULL, `port` varchar(5) NOT NULL default '', `qualify` char(3) default NULL, `restrictcid` char(1) default NULL, `rtptimeout` char(3) default NULL, `rtpholdtimeout` char(3) default NULL, `secret` varchar(80) default NULL, `type` varchar(6) NOT NULL default 'friend', `username` varchar(80) NOT NULL default '', `disallow` varchar(100) default 'all', `allow` varchar(100) default 'g729;ilbc;gsm;ulaw;alaw', `musiconhold` varchar(100) default NULL, `regseconds` int(11) NOT NULL default '0', `ipaddr` varchar(15) NOT NULL default '', `regexten` varchar(80) NOT NULL default '', `cancallforward` char(3) default 'yes', `setvar` varchar(100) NOT NULL default '', PRIMARY KEY (`id`), UNIQUE KEY `name` (`name`), KEY `name_2` (`name`) ) TYPE=MyISAM ROW_FORMAT=DYNAMIC; On Thu, 2006-03-02 at 16:10 -0500, [EMAIL PROTECTED] wrote: Guys, I'm having a hellava time getting realtime to work, focused on sipusers right now, followed the wiki and other examples but still no luck. Using mysql on a seperate server, asterisk actually sees the database and can poll the table realtime load sipusers at the cli but asterisk realtime engine is no pulling the user info. I'm using 1.2.4 stable and have the database info in sip.conf, extconfig.conf and res_mysql.conf. Can anyone using mysql send me sample configs and some insight to getting this going? Thanks. JR ___ --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] check call status during call
Hi, Is there a command (to use in a dial plan), to check the call status during a call. Kind Regards, Arjan Kroon Mobillion B.V. Copernicuslaan 30 Postbus 554 / PO Box 554 6710 BN Ede tel: +31 (0)318-648920 fax: +31 (0)318-648839 mobile: +31 (0)6-55871460 email: [EMAIL PROTECTED] internet: www.mobillion.nl ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Fax Question
[EMAIL PROTECTED] wrote: Hi All, I want to configure fax with Asterisk and I found that we can do this reliably using G711 codec only. Currently my provider is supporting G729 and G711. During the call initiation the call starts with G729 (1'st priority) and Faxing via VoIP is not reliable period. You're only gonna waste time. If you really insist on trying, buy a second DID and register that one with g711 only. Darrick -- Darrick Hartman DJH Solutions, LLC ___ --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] web meetme instructions
Sean Cook wrote: First things first... use the latest version... (that I know of) http://www.fitawi.com/Asterisk/ second... which part are you having problems with? The web piece? or the app_cbmysql? For the app_cbmysql, I have found that the easiest way to work with it is to incorperate it into asterisk-addons... copy app_cbmysql.c into /usr/src/asterisk-addons-1.2.x add MODS+=app_cbmysql.so after MODS=format_mp3... make make clean The web portion is pretty straight forward... just make sure you have register_global=On in php.ini. That being said... I am personally not a big fan of the web portion of the interface ;) I have written my own that allows users to create there own based on their voicemail login. Sean Yes, more detail on your specific problems would be helpful. Also, we have made some fairly major enhancements for a customer, and will be putting those back into the public project shortly. And, we intend to contribute on a regular basis. Let us know more detail and we'll help if we can. Thanks, Mike Clark ___ --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] 'quit' isn't in the CLI's 'help'
When you type 'help' from the CLI, it says nothing about 'quit' - or at least not between 'no debug channel' and 'realtime load'. Google told me about it, and I probably should have guessed, but still... Who do I report this to? Bob McDowell ___ --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] Thinking of moving from pure VoIP to PRI - thoughts?
Hello, For a whole lot of different reasons, I am thinking of moving from pure VoIP (my DID provider gives me SIP access and my termination is SIP too) to PRI (possibly keeping termination in VoIP for long distance). FYI, my business is Hosted PBX...and my end-points will stay SIP. Here is the thing: I don't know much about PRI problems, so what can I expect to have to deal with (except for cost of buying the PRI hardware from Digium and learning how to set it up)? In particular, any comments on the following would be appreciated: 1) Using Asterisk to do Fax-To-Email on PRI: does it work? 2) Can I limit the number of lines that a particular customer (that has, for example 10 Polycom 501) can use at a time? Can this number be different for each customer? What Asterisk functionality can I use for this? 3) CPU load when transferring an incoming call via a PRI to a SIP endpoint, vs SIP-to-SIP transfers Thank you, 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] Re: G729 and Meetme
Sorry. Miss type 'can'. I meant 'cannot' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Martin Joseph Sent: Friday, March 03, 2006 12:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: G729 and Meetme On Mar 2, 2006, at 3:46 PM, Wai Wu wrote: You can really mix G729 encoded frames. So I would guess that licenses are not needed for non-G279 devices. BTW, there is a difference conference app (forgot the name) that only mixes the two parties that have the loudest volumn. It sounds more efficent to me this way. There is no reason to listen to three or more party talking at the same time anyway. I wish this was a joke. Sick and wrong is all I can say. ___ --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] Polling Asterisk for Life
Hello all! On Thu, Mar 02, 2006 at 03:12:05PM -0600, Joseph Tanner wrote: The problem isn't that asterisk isn't running, it's that asterisk is not responding. When asterisk is in this funky state, I can still run asterisk -r from the command line and get access to the CLI. While in the CLI, the only command that asterisk will respond to is exit which drops me back to the shell. If I try to issue a stop now, asterisk just immediately returns to the CLI prompt. It does this for every single command, except for exit. Joseph Tanner I see exactly this behavior too. It occurs on a system using Queue and AgentCallback Login. I have filed a bug report on this - 0006626. In this state it is not longer possible to put a call to a queue, but it is possible to place other calls through *. So * is not totally blocked, just the queues and the CLI. I do have a TDM400 card on one of the machines where it happens, but I have another, bare bone installation with just SIP and IAX2 clients, were I also see it. * 1.2.4. Håkan Källberg pgppTCzzmzDO3.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polling Asterisk for Life
I have not (yet) had one of my bare bones systems lockup.. but they also don't do 400-700 calls a day. The system that does lockup experiences exactly the issues described here which are you can connect to it asterisk -r and issue commands but nothing responds not even stop now... and you have to kill it. When it does this... no one is able to send or receive calls.. and the PRI signaling seems to be down. (Callers inbound get fastbusy). On 3/3/06, Håkan Källberg [EMAIL PROTECTED] wrote: Hello all! On Thu, Mar 02, 2006 at 03:12:05PM -0600, Joseph Tanner wrote: The problem isn't that asterisk isn't running, it's that asterisk is not responding. When asterisk is in this funky state, I can still run asterisk -r from the command line and get access to the CLI. While in the CLI, the only command that asterisk will respond to is exit which drops me back to the shell. If I try to issue a stop now, asterisk just immediately returns to the CLI prompt. It does this for every single command, except for exit. Joseph Tanner I see exactly this behavior too. It occurs on a system using Queue and AgentCallback Login. I have filed a bug report on this - 0006626. In this state it is not longer possible to put a call to a queue, but it is possible to place other calls through *. So * is not totally blocked, just the queues and the CLI. I do have a TDM400 card on one of the machines where it happens, but I have another, bare bone installation with just SIP and IAX2 clients, were I also see it. * 1.2.4. Håkan Källberg ___ --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] Realtime Extensions hint priority
The instructions on the wiki for asterisk Realtime give the extensions schema with the priority field set to be tinyint(4). This of course cannot hold the value 'hint' The question I have, is the solution simply to set the field to varchar(n) as that will then hold 'hint' or any integer value (if you make it big enough) ? Or is there any known places in the realtime integration where the code expects that field to actually be an integer type field. Thanks for any help. - Kevin ___ --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] Autofill phonebook??
Most softphnes (like Idefisk and X-Lite) have a phonebook. Is there a way I can fill those phonebooks with info from the Asterisk server?Amaury Rodríguez http://liberadospucmm.blogspot.com http://groups.msn.com/telematicaPUCMM2002 Go OpenSource And Be FREE!! Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breeze. ___ --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 Realtime voicemail question
Hello there, I'm successfully using Asterisk Realtime to access information about voicemail users from a MySQL database. Now I'd like to read static voicemail information (such as format, serveremail, etc.) also from a database. Is that possible? If so, I'm assuming one would need to attach voicemail to 2 different database tables, one for users and the other for general configuration. Can anyone explain to me how to do that? Best, Leo ___ --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] Program Buttons on Cisco 79xx Phones
Does anyone have a good resource to learn how to program the soft and hard buttons on a Cisco 7940 or 7960 phone? Using SIP Firmwarethanks. ___ --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] Autofill phonebook??
A lot of phones support LDAP for this purpose. I don't think X-Lite does, but I've never checked.On 3/3/06, Amaury Rodriguez [EMAIL PROTECTED] wrote:Most softphnes (like Idefisk and X-Lite) have a phonebook. Is there a way I can fill those phonebooks with info from the Asterisk server? Amaury Rodríguez http://liberadospucmm.blogspot.com http://groups.msn.com/telematicaPUCMM2002 Go OpenSource And Be FREE!! Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breeze. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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: 2 real phone numbers on one SIP account
Hallo! I have problem with incoming calls on 2 phone numbers registered on same SIP provider account. I've tried averything and nothing seems to work. No matter what I do asterisk system refuses differ betwen them and both got connected to the same extensions. I've tride with: registration = num1:[EMAIL PROTECTED]/ext1 registration = num2:pass:[EMAIL PROTECTED]/extt in sip.conf and with making separate sip.conf enteries... always same result. Any clues? thanx, OneHalf ___ --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] Autofill phonebook??
A lot of phones support LDAP for this purpose. I don't think X-Lite does, but I've never checked.On 3/3/06, Amaury Rodriguez [EMAIL PROTECTED] wrote:Most softphnes (like Idefisk and X-Lite) have a phonebook. Is there a way I can fill those phonebooks with info from the Asterisk server? Amaury Rodríguez http://liberadospucmm.blogspot.com http://groups.msn.com/telematicaPUCMM2002 Go OpenSource And Be FREE!! Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breeze. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk silence suppression?
I will try to test your adaptation. How I congfigureto enable VAD? Regards Jsalas -Mensaje original-De: Moises Silva [mailto:[EMAIL PROTECTED]Enviado el: Friday, February 17, 2006 11:26 AMPara: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: Re: [Asterisk-Users] asterisk silence suppression? The patch you saw is not for the stable branch. Salu2 Jsalas Right, but try using this, i adapted it, no guarantees, i have not made tests, just modified it to apply properly, it would be great if some one can test it:http://chewbacca.ivsol.net/asterisk-1.2.1-silence-suppression-4.patchRegardsOn 2/17/06, Rob Lith [EMAIL PROTECTED] wrote: That a phone setting you must set to not supress silence - i.e. in X-Lite/eyeBeam in the advanced settings/audio there is a silence setting.Same for the SNOMs, most phones should have it.RegardsRob On 2/15/06, Dan Elder [EMAIL PROTECTED] wrote: Hi all, I'm getting some noise gate like effects on our sip lines I think I need to disable silence supression, I'm searching docs not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd to the receiving end and I'd like to turn it off if possible... I'm using these Zultys zip2 phones and they dont' have any silence suppression settings, so it seems that I cant' turn it off there.. any leads? Thx as always___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.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] Program Buttons on Cisco 79xx Phones
Cisco has a book about it: http://www.amazon.com/gp/product/1587050609/sr=8-1/qid=1141401215/ref=pd_bbs_1/103-7257053-6939020?%5Fencoding=UTF8 While this isn't specifically about the SIP image, the XML browser is the same. I also use Cisco::IPPhone (http://search.cpan.org/~mrpalmer/Cisco-IPPhone-0.05/IPPhone.pm ) for the backend.On 3/3/06, Kevin Steil [EMAIL PROTECTED] wrote: Does anyone have a good resource to learn how to program the soft and hard buttons on a Cisco 7940 or 7960 phone? Using SIP Firmware…thanks. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo 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] MixMonitor Problems -- sssshh, don't be too loud
I'm running 1.2.4 and just about every call is cut short. I'm using Cisco IP phones as end points. All the outbound calls are routed via SIP through a PRI line attached to a Cisco 2811.. On 3/2/06, Johnathan Corgan [EMAIL PROTECTED] wrote: Gary Richardson wrote: Now it seems that if I'm really loud on a call, MixMonitor stops recording. The wav file stops growing. The log says nothing. When you hang up the call, MixMonitor reports that it is exiting, even though it hasn't been recording since that loud noise. Has anyone experienced such a problem with MixMonitor? Is MixMonitor well tested?I've seen exactly this with MixMonitor in 1.2.1, but I hadn't isolated it to volume issues, just random occurrences.I haven't seen it yet in a week on 1.2.4, but I don't know if the bug isgone or it just hasn't triggered yet.-Johnathan___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Sipura RMA
Anyone have any luck RMAing a Sipura phone since the Cisco take over? Sipura only has support via email or fax to end users and I haven't gotten a response to either for over 2 months. Linksys Support will jump you through all their scripted hoops to resolve your problem (they hope if they speak with a thick enough accent and make repeat the same steps over and over again that you will just give up). After they have done that, they won't give you an RMA number. They tell you to email Sipura...what a joke. ___ --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 version s this??
i am taking overr the administration of an existing production * PBX but i cant seem to find out which version of * this is. When i use the 'show version' coomandat the cli, i get this:Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-08-10 05:57:59 UTCcan anyone tell me which version of * I am useing?or am i getting the command wrong ___ --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] misdn -- zap problem
I've got a problem with chan-misdn I'm using asterisk with a hfcsusb-adapter in nt-mode connected to an isdn-telephone making calls to other internal clients like sip or sccp are without problems if I call into (or receive a call from) the pstn via a zap-channel (Digium E1-card) my outgoing audio from the misdn-device is very choppy. is where anything I can do about 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
[Asterisk-Users] Echo Cancelation on TE110P
On a 55 station install onto a Cox PRI with a TE110P (Polycom 501 phones) a few users are complaiining about echo. According to the users, the echo seems to be phone number dependant. They claim that certain phone numbers have echo while others dont. Are there any tuning parametes like there is for a TDM400 card? Kerry GarrisonDirector of Technical ServicesTech Data Pros - Orange County's Mobile IT Service Provider(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.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
[Asterisk-Users] sprint FNTM(sp?) line
Is anyone using asterisk with the Zapata hardware and a Sprint FNTM(I have seen this spelled fantom) T1 line? I can take calls, but: 1. I don't know the correct way to get ANI/DNIS (*ANI*DNIS*) 2. I cannot place outbound calls. From a custom app on another hardware/software platform I know from experience that to place the outbound call, I need to go offhook and then wait for a wink from the Sprint switch before sending the DTMF digits, but I am relatively new to asterisk and don't know how to accomplish this. I just updated to asterisk-1.2.4 built from source if that helps. Please help! -Greg Lim Confidentiality Notice This message (including any attachments) is intended only for the person or entity to which it is addressed and may contain proprietary or confidential information. If you are not the named addressee, you are not authorized to read, print, retain, copy or disseminate this message or any part of it. If you have received this message in error, please notify the sender immediately and please delete all copies of this message. All e-mail sent to the originating address will be received by the InfiCorp e-mail system and is subject to archiving and review by someone other than the addressee. ___ --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 version s this??
Dumpolid Exeplish wrote: i am taking overr the administration of an existing production * PBX but i cant seem to find out which version of * this is. When i use the 'show version' coomandat the cli, i get this: Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-08-10 05:57:59 UTC can anyone tell me which version of * I am useing?or am i getting the command wrong If it was built from CVS you really should have the source it was built from. Otherwise you need to look carefully at what files are installed(libs for example) and the config files so you can document the existing setup better. ___ --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] G.726 codec - can we select bandwidth?
Can someone please tell me if it's possible to select the G726 codec bandwidth for an IAX trunk between two Asterisk 1.2.5 servers? I can select disallow=all / allow=g726 but I think it defaults to the g726-32 variant. Is there any way of forcing Asterisk to use g726-24 for such a trunk example? Thanks Peter This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ --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 at large
On Thu, Mar 02, 2006 at 04:14:34PM -0700, Douglas Garstang wrote: The best way to achieve maximum manageability is to design a MySQL database and develop AGI scripts (in your language of choice) that work to that design. I've found that it has been far easier to develop complex routing logic in python than it is in the horrible assembler-like Asterisk extensions.conf language. In a perfect world, there would be _ONE_ line in your extensions.conf, and it would be: exten = _.,1,AGI(routing_script) I assume your routing script handles exten = h (which is called when the call disconnects), exten = i (which is handled when an invalid number is dialed), as well as extens a, o and others? Because the above pattern match also matches the extensions I just mentioned. Perhaps you would condsider using exten = _X. that will only match extensions that are all numbers AND are at least 2 digits long. ___ --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] login/logout agents in a specific queue
In Asterisk the Agent / Queue setup is kinda different than most people may expect. You can use a Queue without using Agents and Agents can be used without Queues. Agents however extend normal channels with the ability to login/logout/pause that is not available on Zap/SIP/IAX/etc. I assume that you are using Agent/foo on both queues. Then you will need to dynamically add and remove that agent from the queues using AddQueueMember and RemoveQueueMember. Anything stored in queues.conf will be used when Asterisk is restarted/reloaded, however you can add/remove later as needed. Just keep in mind if you have the agent default to both queues, they remove themselves from one, then you reload Asterisk putting them back in both. Reloading asterisk also undoes pause I've found... --johann nik600 wrote: hi if i have an agents that figure as a member in more than one queue, how can i login / logout him in a specific queue an not in all queues? ___ --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] Sipura RMA
I had the same issue with issue with Sipura. I went though email support. They finaly said email another address to get a RMA. That support made me go though everything the 1st one did again. And in the end they never responded after they decided I needed a RMA. So it never got resolved. Hugh L. Johnson wrote: Anyone have any luck RMAing a Sipura phone since the Cisco take over? Sipura only has support via email or fax to end users and I haven't gotten a response to either for over 2 months. Linksys Support will jump you through all their scripted hoops to resolve your problem (they hope if they speak with a thick enough accent and make repeat the same steps over and over again that you will just give up). After they have done that, they won't give you an RMA number. They tell you to email Sipura...what a joke. ___ --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] G.726 codec - can we select bandwidth?
Whisker, Peter wrote: Can someone please tell me if it's possible to select the G726 codec bandwidth for an IAX trunk between two Asterisk 1.2.5 servers? I can select disallow=all / allow=g726 but I think it defaults to the g726-32 variant. Is there any way of forcing Asterisk to use g726-24 for such a trunk example? Thanks Peter Peter, It think Asterisk only supports g726-32 because it is either very similar or identical to ADPCM. Or something like that. I might be confused. -- Kristian Kielhofner ___ --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] Echo Cancelation on TE110P
only for the whole cardthe tx and rx gain affect all 24 channels. -D From: [EMAIL PROTECTED] on behalf of Kerry Garrison Sent: Fri 3/3/2006 11:19 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Echo Cancelation on TE110P On a 55 station install onto a Cox PRI with a TE110P (Polycom 501 phones) a few users are complaiining about echo. According to the users, the echo seems to be phone number dependant. They claim that certain phone numbers have echo while others dont. Are there any tuning parametes like there is for a TDM400 card? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.techdatapros.com http://www.techdatapros.com/ winmail.dat___ --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] Sipura RMA
Its good for us to post thing about different companies customer service. I feel one of the most important points when buying a product is if the company is going to stand behind it. I had purchased some sipuras that did not work but was lucky and able to send them back to the store I ordered them from. Posts like this make me not want to buy a Sipura product ever again. I think the best way to show sipura that they just can't treat their customers that way is to stop purchasing their products. Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 Karl Davis wrote: I had the same issue with issue with Sipura. I went though email support. They finaly said email another address to get a RMA. That support made me go though everything the 1st one did again. And in the end they never responded after they decided I needed a RMA. So it never got resolved. Hugh L. Johnson wrote: Anyone have any luck RMAing a Sipura phone since the Cisco take over? Sipura only has support via email or fax to end users and I haven't gotten a response to either for over 2 months. Linksys Support will jump you through all their scripted hoops to resolve your problem (they hope if they speak with a thick enough accent and make repeat the same steps over and over again that you will just give up). After they have done that, they won't give you an RMA number. They tell you to email Sipura...what a joke. ___ --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] Problem with NAT!!!
On Fri, 2006-03-03 at 10:45 +0100, serge messa wrote: Hi all I'm a newbie in asterisk.I install asterisk server successfully. I configure this server to traverse NAT. Using Xlite clients, i make a call between 2 local networks through Internet.Asterisk server is installed on a host with public IP. client A (in the LAn A) and client B (in the LAN B) are registered. When i make a call from the LAN A to the LAn B, everything goes well.But, when i try to make a call from the Lan B to the Lan A, the xlite client B, How do connect Lan A and Lan B to the internet? Do they both have a public static IP or are they dynamically assigned? Are they both the same routers? It might be far off but here's a couple of possible reasons: a) either one LAN keeps changing it's public IP (or just bad timing that the IP of Lan A changed when you tried to place your call to it) b) Either Router (a or b) might not allow the relevant packets through to xlite (or to the internet) Can you give more details on your configuration? Can you provide asterisk logs? Conrad ___ --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] TE411P VPM
Aaron Daniel wrote: Thanks :) When we were using Mark2 with aggressive suppression, we had zero problems, but decided to go with the hardware canceler in our new gateway since hardware's supposed to be better than software... hopefully this works for us too. 'aggressive suppression' is half-duplex; no hardware echo canceler is going to operate in that mode. ___ --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] Child PID's
Matt Schulte wrote: All, I'm not sure how to word this question but we're noticing a lot of our asterisk boxes no longer have multiple asterisk child processes. i.e. doing a 'ps ax' reveals only 1 asterisk PID when normally I'm used to seeing 8+ .. There is no rhyme or reason to it, and we're using the safe_asterisk script which has always worked in the past. Ast 1.2.4, zap 1.2.4, naturally.. This is completely normal; if your distro is using LinuxThreads, then you will see multiple processes, but if it is using NPTL (as current distros do), then you will only see one process because the threads are not shown as processes. ___ --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] web meetme instructions
Jordan- I'm not sure if you found the files and instructions on www.fitawi.com/Asterisk/. If you did I can offer you a full refund of the purchase price. (oh right, it's free, I forgot) I'm afraid I did make some assumptions about which packages are installed on the system. If you can be specific about which problems you had, I can address them for future updates and other users. I've tried to cover what I thought were the points that would not be obvious to the installer. Again, if you can identify specific issues I can help with them and make the documentation better. Sean- Thanks for the comments and tips about -Addons. That's been on my ToDo list for a while. I'll intergrate your comments into the Readme. Mike- I'm looking forward to your additions. We've been using it for a month in production, without any complaints. Even my ugly little php script to announce the end of a ongoing conference has been stable for that period of time. All- I've seen an increased interest in this package over the last couple of months, yet I have not made any new announcements recently. I'm curious how people have found it, and if there is an Asterisk related news site that I missed. Dan ___ --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] Child PID's
On Fri, Mar 03, 2006 at 11:39:49AM -0600, Kevin P. Fleming wrote: Matt Schulte wrote: All, I'm not sure how to word this question but we're noticing a lot of our asterisk boxes no longer have multiple asterisk child processes. i.e. doing a 'ps ax' reveals only 1 asterisk PID when normally I'm used to seeing 8+ .. There is no rhyme or reason to it, and we're using the safe_asterisk script which has always worked in the past. Ast 1.2.4, zap 1.2.4, naturally.. This is completely normal; if your distro is using LinuxThreads, then you will see multiple processes, but if it is using NPTL (as current distros do), then you will only see one process because the threads are not shown as processes. here is where my show threads command would come handy :) http://bugs.digium.com/view.php?id=6053 *CLI show threads 0x8940800 autoservice_run started at [ 115] autoservice.c ast_autoservice_start() 0x8897400 do_monitor started at [ 6877] chan_zap.c restart_monitor() 0x8897000 do_monitor started at [ 3038] chan_skinny.c restart_monitor() 0x8931c00 accept_threadstarted at [ 3223] chan_skinny.c reload_config() 0x8931800 do_monitor started at [11168] chan_sip.c restart_monitor() 0x888f800 sound_thread started at [ 1465] chan_oss.c store_config() 0x888f400 do_monitor started at [ 3548] chan_mgcp.c restart_monitor() 0x87ca800 network_thread started at [ 8327] chan_iax2.c start_network_thread() 0x87ca400 sched_thread started at [ 8326] chan_iax2.c start_network_thread() 0x87ca000 iax2_process_thread started at [ 8317] chan_iax2.c start_network_thread() 0x86c1c00 iax2_process_thread started at [ 8317] chan_iax2.c start_network_thread() 0x86c1800 iax2_process_thread started at [ 8317] chan_iax2.c start_network_thread() 0x86c1400 iax2_process_thread started at [ 8317] chan_iax2.c start_network_thread() 0x86c1000 iax2_process_thread started at [ 8317] chan_iax2.c start_network_thread() 0x8576c00 iax2_process_thread started at [ 8317] chan_iax2.c start_network_thread() 0x8576800 iax2_process_thread started at [ 8317] chan_iax2.c start_network_thread() 0x8576400 iax2_process_thread started at [ 8317] chan_iax2.c start_network_thread() 0x8576000 iax2_process_thread started at [ 8317] chan_iax2.c start_network_thread() 0x8229c00 iax2_process_thread started at [ 8317] chan_iax2.c start_network_thread() 0x8228800 scan_thread started at [ 424] pbx_spool.c load_module() 0x8228400 process_precache started at [ 2142] pbx_dundi.c start_network_thread() 0x8228000 network_thread started at [ 2141] pbx_dundi.c start_network_thread() 0x813a800 do_parking_threadstarted at [ 2052] res_features.c load_module() 0x813a400 do_devstate_changes started at [ 269] devicestate.c ast_device_state_engine_init() 0x813a000 listener started at [ 713] asterisk.c ast_makesocket() 25 threads listed. cheers luigi --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] Hardware Requirements for 1M minutes
I'm doing an install for a client with the following requirements. - 1 Million minutes of outbound calling - Calls come in to asterisk via SIP/IAX and terminated to third party provider via SIP - Codec usage will be about 70% g711 30% g729 (there should be no transcoding) - 100% IP setup with no voice cards in the box They have a box on hand with a single 3.2ghz P4 w/Hyper-threading, 2GB RAM Dual 10/100 card. The question is... Will their current system be OK for them? If not, what would you recommend? I realize I may be leaving out some needed info, hopefully this is enough to go on. regards, David ___ --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] [HELP] dial plan continue for outbound channel on disconnect
Does anyone know if there is a way to continue in the dial plan for the
called (outbound) channel if the caller channel disconnects? Something
like this:
*
[call_client]
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:1},30,g)
exten = _9NXX,2,Playback(some_file)
exten = _9NXX,3,Hangup
*
I am trying to do a auto call routing from an applications with out having
to determine the channel inuse and using a AMI redirect or a dtmf transfer
code.
Anyone have any experience with this or is there an better/easier way to
accomplish this.
___
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[Asterisk-Users] dtmf tones problem with unicall and E1
Guys. I have a te100p with unicall and an E1 and Im having problem with DTMF tones but the weird thing is, I only have problems sending the tones to certain phone numbers, anybody seen this behavior? Asterisk shows on the console the dtmf tone been pressed but seems the other side is not getting them, and this just happens with certain phone numbers, not all.. Any clues(tips? ___ --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] Lowering Server Load
Hi Ron! Well, I dont have raid involved here but do use SATA. I have 14 sip phones and an average of 5 calls at a time. I was thinking about recording in gsm to a ramdrive and then copying the files to the disk at certain intervals. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McCarthySent: Thursday, March 02, 2006 9:15 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Lowering Server Load Also, SATA on a onboard SATA card will eat more CPU then a SCSI system. Are you running software RAID by chance with your SATA? SCSI or SCSI Raid will not each CPU near as much since the HBA does all the work and does tie up the CPU with all its I/O's. We have successfulyl recorded 5+ calls at a time via dual xeon 3.0 with 10K SCSI drives in RAID-5 with no issuses running about 30 PRI channels and anywhere from 50-75 SIP channels, all with g729 encoding.Hope this helps!Ron On 3/2/06, Anton Krall [EMAIL PROTECTED] wrote: Yep, I tried it and indeed, it lowers cpu usage, so I switched from wav togsm format and Im thinking about doing the ramdisk solution for recording...Sounds like a good move?|-Original Message-|From: [EMAIL PROTECTED]|[mailto:[EMAIL PROTECTED]] On Behalf Of |Matt Riddell [NZ]|Sent: Thursday, March 02, 2006 2:04 AM|To: Asterisk Users Mailing List - Non-Commercial Discussion|Subject: Re: [Asterisk-Users] Lowering Server Load||Can you try not recording for a bit and see if that helps? ||--|Cheers,||Matt Riddell|___||http://www.sineapps.com/news.php (Daily Asterisk News - html)|http://freevoip.gedameurope.com (Free Asterisk Voip Community) |http://www.sineapps.com/rssfeed.php (Daily Asterisk News -|rss) ___|--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 listTo 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] what version s this??
Dumpolid Exeplish wrote: i am taking overr the administration of an existing production * PBX but i cant seem to find out which version of * this is. When i use the 'show version' coomandat the cli, i get this: Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on 2005-08-10 05:57:59 UTC This is the development version of Asterisk as it was on 2005-08-10 05:57:59 UTC. There is no actual version number, since this is the development version. ___ --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] Same CID on multiple users(friends9 in SIP.conf
Michiel van Baak wrote: On 13:02, Wed 01 Mar 06, Arne Morten Johansen wrote: Hi there. Is it possible to have different sip users have the same CallerId number in sip.conf. I need this because we got multiple companies on this Asterisk box. Company A's internal numbers: CID: User: 1000 - User 1 2000 - User 2 3000 - User 3 4000 - User 4 Company B's internal numbers: CID: User: 1000 - User 5 2000 - User 6 3000 - User 7 4000 - User 8 Hi, This is possible, but you have to add the sip users in sip.conf with a unique name. We do it like this: [1002_1000] where 1002 is the companies customer number and 1000 is their internal cid in extensions.conf you make a context for every company. Cheers, We use the MAC address for the [whatever]. That way we never, ever think device = extension. ___ --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] IAX2 register problem
Hi guys, I am trying to register IP IAX2 phone to our Asterisk server. this is what I see on traffic debug between the asterisk server and IP phone. I do not see anything in asterisk console. Can somebody give me hints what could be the reason that phone is not registering? Thank you in advance. Bart 678.878478 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 2, timestamp 3ms ACK680.576003 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 3, timestamp 20020ms PING680.576241 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 3, timestamp 20023ms LAGRQ681.652360 200.2.165.139 - 62.204.64.161 IAX2 IAX, source call# 31736, timestamp 3ms REGREQ681.653112 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 3, timestamp 3ms ACK682.580584 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 3, timestamp 20020ms PING682.580842 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 3, timestamp 20023ms LAGRQ682.580920 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 3, timestamp 10024ms LAGRQ682.581233 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 3, timestamp 17ms REGAUTH682.874132 200.2.165.139 - 62.204.64.161 IAX2 IAX, source call# 31738, timestamp 3ms REGREQ682.874809 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 2, timestamp 3ms ACK683.760566 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 2, timestamp 10004ms LAGRQ685.671308 200.2.165.139 - 62.204.64.161 IAX2 IAX, source call# 31736, timestamp 3ms REGREQ685.671649 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 3, timestamp 3ms ACK685.760539 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 2, timestamp 10004ms LAGRQ685.760842 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 2, timestamp 3ms REGAUTH686.882317 200.2.165.139 - 62.204.64.161 IAX2 IAX, source call# 31738, timestamp 3ms REGREQ686.882763 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 2, timestamp 3ms ACK689.639694 200.2.165.139 - 62.204.64.161 IAX2 IAX, source call# 31736, timestamp 3ms REGREQ689.640265 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 3, timestamp 3ms ACK690.574225 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 3, timestamp 30018ms LAGRQ692.576738 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 3, timestamp 30018ms LAGRQ692.580576 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 3, timestamp 20020ms PING692.580745 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 3, timestamp 20023ms LAGRQ692.580886 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 3, timestamp 10024ms LAGRQ693.759809 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 2, timestamp 20003ms PING693.759912 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 2, timestamp 20006ms LAGRQ694.877138 200.2.165.139 - 62.204.64.161 IAX2 IAX, source call# 31738, timestamp 3ms REGREQ694.877609 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 2, timestamp 3ms ACK695.760621 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 2, timestamp 20003ms PING695.760842 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 2, timestamp 20006ms LAGRQ695.761009 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 2, timestamp 10004ms LAGRQ695.761323 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 2, timestamp 3ms REGAUTH698.889998 200.2.165.139 - 62.204.64.161 IAX2 IAX, source call# 31738, timestamp 3ms REGREQ698.890435 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 2, timestamp 3ms ACK702.898555 200.2.165.139 - 62.204.64.161 IAX2 IAX, source call# 31738, timestamp 3ms REGREQ702.916617 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 2, timestamp 3ms ACK703.678273 200.2.165.139 - 62.204.64.161 IAX2 IAX, source call# 31739, timestamp 3ms REGREQ703.678978 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 13, timestamp 2ms REGAUTH703.760686 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 2, timestamp 30004ms LAGRQ703.761573 200.2.165.139 - 62.204.64.161 IAX2 IAX, source call# 31739, timestamp 3ms REGREQ703.761871 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 13, timestamp 3ms ACK704.019989 200.2.165.139 - 62.204.64.161 IAX2 IAX, source call# 31739, timestamp 3ms REGREQ704.020678 62.204.64.161 - 200.2.165.139 IAX2 IAX, source call# 13, timestamp 3ms ACK ___ --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] ignore a DID?
Jesse Guardiani wrote: Hello, What is the best way to ignore a DID and not pick up the line? I don't want to incur charges on the line (T1 PRI), so would Hangup pick up the line, then hang up? Or can I use Hangup? Use the Congestion application. ___ --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] my zap channel not ringing
ADEGOKE ARUNA wrote: I need your help I have a sangoma A104D on my dell server; I got card status ok with no alarm If I dialed the extension 6210006, it shows the output as stated below, but there is no ringing from the pstn number nor the iax softphone am using on my pc. I will be glad if someone can give me a working config? What I want to achieve is to send all my call to the pstn on A104D? The pstn am talking to is alcatel S12 and the pri status on their switch is showing the channel is external blocked meaning that the channels are blocked from my asterisk box. . Output from asterisk cli -- Accepting AUTHENTICATED call from 10.80.1.151: requested format = ulaw, requested prefs = (), actual format = gsm, host prefs = (), priority = mine -- Executing Answer(IAX2/marko-3, ) in new stack -- Executing Dial(IAX2/marko-3, Zap/g1/6210006,60,r) in new stack -- Called g1/6210006,60,r -- Zap/1-1 answered IAX2/marko-3 -- Hungup 'Zap/1-1' == Spawn extension (default, 6210006, 2) exited non-zero on 'IAX2/marko-3' -- Hungup 'IAX2/marko-3' A classic fix for no ringback after an answer is to make sure you have a /etc/asterisk/indications.conf. I don't know if that's your problem, but it's an easy thing to try. ___ --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] Lowering Server Load
way to go Matt! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt RothSent: Thursday, March 02, 2006 11:51 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Lowering Server Load All,Just a quick update on our progress with the RAM disk solution for digitally recording large numbers of calls via Monitor. We are currently recording approximately 80 - 100 concurrent calls to the PCM format on our production server. We also have over 220 dynamic agents logged into 10 queues handling calls across 4 offices (1 local, 3 remote). All of our calls are SIP to SIP (a Cisco AS5400 terminates our Ts) using the u-Law codec and we do no transcoding or DSP on the Asterisk box. Yesterday, a total of over 8300 calls were handled. The box is running roughly 77% - 80% idle.As we add more clients to the box, I'll update the list with the results.For more details of our setup see here http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/118497 and here http://lists.digium.com/pipermail/asterisk-users/2005-October/127919.html.Matthew RothInterMedia Marketing SolutionsSoftware Engineer and Systems DeveloperRon McCarthy wrote: Also, SATA on a onboard SATA card will eat more CPU then a SCSI system. Are you running software RAID by chance with your SATA? SCSI or SCSI Raid will not each CPU near as much since the HBA does all the work and does tie up the CPU with all its I/O's. We have successfulyl recorded 5+ calls at a time via dual xeon 3.0 with 10K SCSI drives in RAID-5 with no issuses running about 30 PRI channels and anywhere from 50-75 SIP channels, all with g729 encoding.Hope this helps!Ron On 3/2/06, Anton Krall [EMAIL PROTECTED] wrote: Yep, I tried it and indeed, it lowers cpu usage, so I switched from wav togsm format and Im thinking about doing the ramdisk solution for recording...Sounds like a good move? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk management interface
I was talking about the web managers posted, if they were free,gpl,gnu. Some are commercial, but the first ones posted looked very nice and I think they are free, but was asking to the poster. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Martin Joseph |Sent: Thursday, March 02, 2006 11:14 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] asterisk management interface | | |On Mar 2, 2006, at 12:32 PM, Anton Krall wrote: | | Looks very nice.. Is it GPL, GNU? | |Maybe if you trimmed you posts and pasted relevant quotes, we |could have some idea what this question means... | |___ |--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: Asterisk at large
Oops. Yes, Actually I do have _X. I didn't copy and paste what I had, I just typed it from memory. Thanks for the tip. -Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Friday, March 03, 2006 9:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Asterisk at large On Thu, Mar 02, 2006 at 04:14:34PM -0700, Douglas Garstang wrote: The best way to achieve maximum manageability is to design a MySQL database and develop AGI scripts (in your language of choice) that work to that design. I've found that it has been far easier to develop complex routing logic in python than it is in the horrible assembler-like Asterisk extensions.conf language. In a perfect world, there would be _ONE_ line in your extensions.conf, and it would be: exten = _.,1,AGI(routing_script) I assume your routing script handles exten = h (which is called when the call disconnects), exten = i (which is handled when an invalid number is dialed), as well as extens a, o and others? Because the above pattern match also matches the extensions I just mentioned. Perhaps you would condsider using exten = _X. that will only match extensions that are all numbers AND are at least 2 digits long. ___ --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] my zap channel not ringing
Thank you , but the pstn subscriber am calling is not ringing at all But I can here ringing from my own softphone from zap channel. Thankx -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Friday, March 03, 2006 7:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] my zap channel not ringing ADEGOKE ARUNA wrote: I need your help I have a sangoma A104D on my dell server; I got card status ok with no alarm If I dialed the extension 6210006, it shows the output as stated below, but there is no ringing from the pstn number nor the iax softphone am using on my pc. I will be glad if someone can give me a working config? What I want to achieve is to send all my call to the pstn on A104D? The pstn am talking to is alcatel S12 and the pri status on their switch is showing the channel is external blocked meaning that the channels are blocked from my asterisk box. . Output from asterisk cli -- Accepting AUTHENTICATED call from 10.80.1.151: requested format = ulaw, requested prefs = (), actual format = gsm, host prefs = (), priority = mine -- Executing Answer(IAX2/marko-3, ) in new stack -- Executing Dial(IAX2/marko-3, Zap/g1/6210006,60,r) in new stack -- Called g1/6210006,60,r -- Zap/1-1 answered IAX2/marko-3 -- Hungup 'Zap/1-1' == Spawn extension (default, 6210006, 2) exited non-zero on 'IAX2/marko-3' -- Hungup 'IAX2/marko-3' A classic fix for no ringback after an answer is to make sure you have a /etc/asterisk/indications.conf. I don't know if that's your problem, but it's an easy thing to try. ___ --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] SIP Problem - Asterisk to Provider Gateway
Hi All, I'm stumped on a weird problem. I have an * server working fine for local SIP phones and IAX2 connections. We just provisioned a second Ethernet port to attach to a local SIP provider. PSTN calls incoming work fine: PSTN - SIP Provider - SIP - * but outgoing calls are not. Call setup takes place and the caller can hear about 1-2 seconds of audio before the SIP provider cancels the call and sends back a BYE message. They haven't made any changes on their end (metaswitch). The wierd part is that yesterday I was having the exact opposite problem (outgoing working fine, incoming calls no audio). RTP setup was correct, but * wasn't responding to the RTP packets. Recompiled asterisk with PRI support for the X100P card installed: make make install libpri (1.2.2) make clean make make install zaptel (1.2.3) make clean make make install asterisk (1.2.4) Set zaptel and zapata for the X100P and TDM400P cards (not in use, but using for clock) and the incoming audio was fixed, outgoing not so much. Here is a debug of the SIP session. The ones I'm curious about are the provider OK packets and *'s ACK response. It appears that the SIP provider isn't seeing them. Also, the ACK response time is less than 1ms (with qualify on, the SIP peer quals at 4-6ms). Any assistance would be appreciated. tethereal: 1 0.00 10.70.0.92 - 10.70.0.89 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 2 0.003542 10.70.0.89 - 10.70.0.92 SIP Status: 100 Trying 3 1.214914 10.70.0.89 - 10.70.0.92 SIP/SDP Status: 183 Session Progress, with session description 4 1.216377 10.70.0.89 - 10.70.0.92 SIP Status: 180 Ringing 5 1.528401 10.70.0.89 - 10.70.0.92 SIP/SDP Status: 200 OK, with session description 6 1.528820 10.70.0.92 - 10.70.0.89 SIP Request: ACK sip:[EMAIL PROTECTED]:5060;maddr=10.70.0.89;transport=udp 7 1.771613 10.70.0.89 - 10.70.0.92 SIP/SDP Status: 200 OK, with session description 8 1.772038 10.70.0.92 - 10.70.0.89 SIP Request: ACK sip:[EMAIL PROTECTED]:5060;maddr=10.70.0.89;transport=udp 9 2.271674 10.70.0.89 - 10.70.0.92 SIP/SDP Status: 200 OK, with session description 10 2.272098 10.70.0.92 - 10.70.0.89 SIP Request: ACK sip:[EMAIL PROTECTED]:5060;maddr=10.70.0.89;transport=udp 11 3.271984 10.70.0.89 - 10.70.0.92 SIP/SDP Status: 200 OK, with session description 12 3.272384 10.70.0.92 - 10.70.0.89 SIP Request: ACK sip:[EMAIL PROTECTED]:5060;maddr=10.70.0.89;transport=udp 13 3.522590 10.70.0.89 - 10.70.0.92 SIP Request: BYE sip:[EMAIL PROTECTED];transport=udp 14 3.522947 10.70.0.92 - 10.70.0.89 SIP Status: 200 OK And a few of the sip debug messages for the SIP/SDP and SIP Request ACK packets: -- SIP read from 10.70.0.89:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.70.0.92:5060;branch=z9hG4bK1f0a40b1;rport=5060 From: 4414964319 sip:[EMAIL PROTECTED];tag=as75a2b003 To: sip:[EMAIL PROTECTED];tag=SD6q75199-quantum1.quantum.bm+1+acb61+6a3a58ac Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: DC-SIP/2.0 Allow-Events: message-summary Allow-Events: refer Supported: 100rel Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: REGISTER Allow: OPTIONS Allow: PRACK Allow: UPDATE Allow: SUBSCRIBE Allow: NOTIFY Allow: REFER Accept-Encoding: identity Accept: application/sdp Accept: application/simple-message-summary Contact: sip:[EMAIL PROTECTED]:5060;maddr=10.70.0.89;transport=udp Content-Length: 119 Content-Type: application/sdp v=0 o=- 3244118288 3244118288 IN IP4 10.70.0.89 s=- c=IN IP4 10.70.0.89 t=0 0 m=audio 9196 RTP/AVP 0 a=ptime:20 --- (27 headers 7 lines)--- Found RTP audio format 0 Peer audio RTP is at port 10.70.0.89:9196 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Transmitting (no NAT) to 10.70.0.89:5060: ACK sip:[EMAIL PROTECTED]:5060;maddr=10.70.0.89;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.70.0.92:5060;branch=z9hG4bK7713b672;rport From: 4414964319 sip:[EMAIL PROTECTED];tag=as75a2b003 To: sip:[EMAIL PROTECTED];tag=SD6q75199-quantum1.quantum.bm+1+acb61+6a3a58ac Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI -- SIP read from 10.70.0.89:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.70.0.92:5060;branch=z9hG4bK1f0a40b1;rport=5060 From: 4414964319 sip:[EMAIL PROTECTED];tag=as75a2b003 To: sip:[EMAIL PROTECTED];tag=SD6q75199-quantum1.quantum.bm+1+acb61+6a3a58ac Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Server: DC-SIP/2.0 Allow-Events: message-summary Allow-Events: refer Supported: 100rel Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Allow: REGISTER Allow: OPTIONS Allow: PRACK Allow: UPDATE Allow: SUBSCRIBE Allow: NOTIFY Allow: REFER Accept-Encoding: identity Accept: application/sdp Accept: application/simple-message-summary Contact: sip:[EMAIL
Re: [Asterisk-Users] TE411P VPM
Yep :) We were using that before we got the hardware cancelers in. Aaron Kevin P. Fleming wrote: Aaron Daniel wrote: Thanks :) When we were using Mark2 with aggressive suppression, we had zero problems, but decided to go with the hardware canceler in our new gateway since hardware's supposed to be better than software... hopefully this works for us too. 'aggressive suppression' is half-duplex; no hardware echo canceler is going to operate in that mode. ___ --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] Problem with HT-286 BT-101
Hello all, I am new to [EMAIL PROTECTED] and am having a strange issue with both my new Grandstream HT-286 BT-101. The issue is as follows: Example is with BT-101 (HT-286 shows same behavior) 1) Device registers to Asterisk 2) I can place a call via the BT-101 out my Zap or SIP provider 3) Conversation takes place (yay!) 4) I hang up BT-101 5) BT-101 will no longer dial out until: a) I give asterisk a restart now b) I place a call from another extension in my home TO the BT-101, answer BT-101, hang up BT-101, and all is well for another single outbound call Nothing is logged via sip debug peer 7213 when the phone will not dial. After I reset it above, everything looks/works fine. This behavior also occurs with internal extension to extension calls between the BT-101 (7213) and HT-286 (7214). To me, it sounds like I have something incorrect with my extension configuration (teardown?) for both the BT-101 and HT-286, however, I also have 2 X-Lite softphones, with identical extension configurations as the Grandstream devices, and both of the softphones work flawlessly, and have for several weeks now. Here are my configs. First X-Lite softphone: [7211] username=7211 type=friend record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 7211 BT-101 (firmware 1.0.8.16): [7213] username=7213 type=friend record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device 7213 Please let me know if you need any more of my configurations; any and all help would be appreciated. Thank you! ___ --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] Brainstorming dual-core and Asterisk
Jim Van Meggelen wrote: Let me run something that's been floating about in my noggin by everyone: Given that Asterisk does not make use of dual core CPUs or dual processors... Jim, That statement bothered me, because we are running Asterisk on a multi-processor system to help accomplish our scalability goals. I did some double-checking on it by talking to Matt O'Gorman of Digium. Here is what he had to say: Asterisk makes use of both processors for 99% of things. There are some things like IAX parser or SIP parser that only run on one thread (although Mark [Spencer] recently did multi-threaded IAX), but the heavy stuff like each call spawns a new thread and Linux being awesome like it is will share the load across processors. I mean just run top, you will see load should be fairly balanced. On our production server we are currently handling ~90 concurrent calls with digital recording via Monitor as well as ~200 dynamic agents logged in. top shows us running around 80% idle with processor 0 hovering around 70% idle, and processors 1, 2, and 3 around 85%. Your VMWare idea is very interesting, but I think it's unnecessary. I believe that Asterisk *does* perform better with HyperThreading/logical processors disabled. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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: TDM400P digium card
Would QoS on a managed switch solve the ARP problem? I'm not sure about QoS, because we haven't tried it, but my initial feeling is probably not. We solved our problem by separating the network segments completely, which provides us with better security as well as the quality we required. I say probably not because it wasn't a case of the link segments being saturated with ARP packets. There was plenty of overhead left to handle the voice, on any particular segment a phone was on. This is not hard at all, for ulaw over a 10MB link, for a single phone. The problem was that the phones themselves seemed to be spending far too much effort ignoring the ARP packets which didn't belong to them. The symptom we saw was unreasonably high decode latency on the phone status page (half a second or more, and not stable). Jitter and transit latency measurements taken at the phone were not problems at all. Alan Ferrency pair Networks, Inc. [EMAIL PROTECTED] On Wed, 1 Mar 2006, mustardman29 wrote: Would QoS on a managed switch solve the ARP problem? Regarding sound quality issues with Sipura SPA-841 phones... snip After that: Are your phones sharing the same network segments as your non-VoIP ethernet data? Do you have a lot of ethernet traffic? We found that even on a fully switched network, if the SPA-841's received excessive ARP traffic (which is broadcast to all switch segments, even though most other network packets are suppressed), we had periodic robot voice sound issues. ___ --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 coder conflicts
We have an external FXO/FXS, and use Asterisk as a call router. We want to use G723 for the actual phone calls, because we have limited bandwidth on our return direction. This has been working fine so far. However, now we want to set up Asterisk to handle PBX menues and accept extentions. Asterisk, of course, uses GSM for its messages, and cannot terminate G723 calls. So I want to tell Asterisk, FXO, and FXS to use GSM for messages and G723 for the data connection. The FXO/FXS would support this, but Asterisk isn't working as I wish. Though I can provide it with a list of coding schemes: ; set up codecsdisallow=allallow=gsmallow=g723.1allow=g729 It never uses anything but the first one. So if I use the above scheme, messages are played successfully, but the calls go through using GSM. If I put G723.1 first, Asterisk aborts with an error message (cannot convert gsm to g723.1). Is there some way I can solve this problem, such as explicitly telling * to use GSM for its messages, and G723 for the handed-off data connection?? ___ --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] Bad quality between SIP and TDM
Hi. Im facing a really bad voice quality when a make calls between a tdm user and a sip user. Take a look at the following scenario: sip-user asterisk TDM22B(fxo) PABX and PABX my-tdm-extension When the sip-user places a call to my-tdm-extension, the call goes through the TDM22B followed by the PABX and then I answer it in my-tdm-extension. For the sip user the quality of the voice is normal, but for the tdm-extension its unacceptable. I got some sequences of choppy/picotted voice. The invert situation is also true, even if the tdm-extension place the call to the sip user, the voice also is terrible for de tdm side. First it looked like a problem with bandwidth but calls between the sip-user and another sip user (this another sip is in the same building that the tdm-extension is) are excellent, so this tells me that bandwidths isnt my problem. My PABX extensions group work pretty well among them selves so look like that isnt the problem either. I really think it is something with IRQ misses or some bus problem but Ive already followed the steps mentioned in voip-info.org to test IRQ misses and Im still unable to figure out what is the problem. Im using the GSM codec on the sip-user, but even with ulaw the problem persists. Any help would be appreciated. Thanks, Filipe Mordhorst smime.p7s Description: S/MIME cryptographic 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
[Fwd: Re: [Asterisk-Users] problem with incoming peer (cisco as5400)]
For the archives record. Original Message Subject:Re: [Asterisk-Users] problem with incoming peer (cisco as5400) Date: Fri, 03 Mar 2006 11:14:50 -0600 From: Miguel [EMAIL PROTECTED] To: Ron McCarthy [EMAIL PROTECTED] References: [EMAIL PROTECTED] [EMAIL PROTECTED] Ron McCarthy wrote: Hi Miguel, I wish I knew what was going on your setup, but I have a question for ya :) On your 5450 do you have just have T1 termianted to it? Has this been working well until now? We are looking into this im trying to see if it work well... It looks from your configs your using SIP instead of H.323 from Cisco - * ? Any help on this would be great! Thanks Ron Ron , this is the picture E1(S7) E1(isdn)SIP(ethernet) PSTN (Switch DMS300) --- AS5400 --- * (1.0) Sip phones This setup has beeen working very well, beautifully if you ask me :-) Sadly, this is not working anymore after the upgrade to 1.2.4, and it seems nobody has an explanations for this change. regards Miguel ___ --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] Sipura RMA
Anyone have any luck RMAing a Sipura phone since the Cisco take over? Sipura only has support via email or fax to end users and I haven't gotten a response to either for over 2 months. Sipura's policy was to handle RMAs through resellers. Since taking over, Linksys appears to have maintained the same policy for the SPA- devices. So, contact your reseller. BTW I always get quick responses from [EMAIL PROTECTED] (which I believe is forwarded to [EMAIL PROTECTED]). Nabeel ___ --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] Sipura RMA
I have had the same issue with Sipura as well, they gave me quite the run around on 1 bad 2002 ATA and decided to just forget about it and bought a new one to save time. On 3/3/06, Michael Sampson [EMAIL PROTECTED] wrote: Its good for us to post thing about different companies customer service. I feel one of the most important points when buying a product is if the company is going to stand behind it. I had purchased some sipuras that did not work but was lucky and able to send them back to the store I ordered them from. Posts like this make me not want to buy a Sipura product ever again. I think the best way to show sipura that they just can't treat their customers that way is to stop purchasing their products. Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 Karl Davis wrote: I had the same issue with issue with Sipura. I went though email support. They finaly said email another address to get a RMA. That support made me go though everything the 1st one did again. And in the end they never responded after they decided I needed a RMA. So it never got resolved. Hugh L. Johnson wrote: Anyone have any luck RMAing a Sipura phone since the Cisco take over? Sipura only has support via email or fax to end users and I haven't gotten a response to either for over 2 months. Linksys Support will jump you through all their scripted hoops to resolve your problem (they hope if they speak with a thick enough accent and make repeat the same steps over and over again that you will just give up). After they have done that, they won't give you an RMA number. They tell you to email Sipura...what a joke. ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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] Echo Cancelation on TE110P
It is my understanding that when you hear echo the problem is on the other end. So if a caller complains they hear echo that is something you should be dealing with, but if you hear echo that is the phone companies fault. Now with a normal phone, the phone company will only echo cancel long distance calls. For local calls the latency is not high enough to matter. But with VOIP the added latency creates echo even for local calls. I think the reason you hear it on some numbers and not others is that the phone companies are doing echo cancel on some of those calls and not on others. Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 Kerry Garrison wrote: On a 55 station install onto a Cox PRI with a TE110P (Polycom 501 phones) a few users are complaiining about echo. According to the users, the echo seems to be phone number dependant. They claim that certain phone numbers have echo while others dont. Are there any tuning parametes like there is for a TDM400 card? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949)502-7819 x200- [EMAIL PROTECTED] http://www.techdatapros.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 ___ --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] Sipura RMA
Its good for us to post thing about different companies customer service. I feel one of the most important points when buying a product is if the company is going to stand behind it. I had purchased some sipuras that did not work but was lucky and able to send them back to the store I ordered them from. Posts like this make me not want to buy a Sipura product ever again. I think the best way to show sipura that they just can't treat their customers that way is to stop purchasing their products. FWIW, there is a very high probability that Sipura will disappear from the market since the primary asset (eg, software) has been sold to Linksys. The Linksys folks are the bigger fish in the pond and will swallow up the smaller fish. You've already seen the first phase of that happen, and it's my believe the next phase is in progress. ___ --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] dtmf tones problem with unicall and E1
On Mar 3, 2006, at 9:48 AM, Anton Krall wrote: Guys. I have a te100p with unicall and an E1 and Im having problem with DTMF tones but the weird thing is, I only have problems sending the tones to certain phone numbers, anybody seen this behavior? Asterisk shows on the console the dtmf tone been pressed but seems the other side is not getting them, and this just happens with certain phone numbers, not all.. I have seen this through my FXO, when the transmitted volume is too loud and apparently the audio breaks up at the other end? This was somewhat speculation on my part, but adjusting the transmit gain down did seem to resolve it, so that was the proof. It's pretty difficult with such a huge variety of different phone systems and equipment out there to get them all working acceptably. Of course my biggest issues have been with the stupid phone system at my wife's workplace! I had to retune my gains for that system after I thought I was done... It's clearly a compromise, and I suppose if the hardware (FXO in my case) had a GOOD auto gain adjust that might help... Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Fax Question
Your best bet is to just not use fax machines. They are outdated technology. With email there is little reason to use fax machines anymore. But for some reason people just feel the need to hang on to them. A good solutions is to get a fax machine that supports fax to email. We have a Brother 8440D, that you can type in a fax number or a email address and hit send, works the same both ways. Only with the email it attaches a tiff image to an email. You can also set up an email box that the fax machine will check and download and print faxes from. And if you must have a fax number, you can sign up for a fax to email phone number for a provider like efax.com. Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 Darrick Hartman wrote: [EMAIL PROTECTED] wrote: Hi All, I want to configure fax with Asterisk and I found that we can do this reliably using G711 codec only. Currently my provider is supporting G729 and G711. During the call initiation the call starts with G729 (1'st priority) and Faxing via VoIP is not reliable period. You're only gonna waste time. If you really insist on trying, buy a second DID and register that one with g711 only. Darrick ___ --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] snom 320 MWI light
I am using a snom 320 running 5.3.6 with Asterisk 1.2.4. In the sip.conf entry, I have [EMAIL PROTECTED] and vmexten=*98. The light on the snom 320 turns on when I have voicemail and the retrieve button dials the correct extensions. However, the light turns off immediately after making the call to voicemail, even if I do not check the voicemail. FYI Received the following from a vendor: Currently there is not a way to keep the MWI light to stay on after hitting retrieve button on the Snom. The best option at this point is to set checkmwi=1 in the general section of your sip.conf file. This will cause the light to turn back on shortly if there are un-checked messages waiting. ___ --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] 160 analogue phones..
I guess if I was going to do this I would either have a sip adapter at each phone. Or have to * boxes. One is connected to the PRIs. Then connected to that via an IAX2 trunk is another asterisk box that is full of the 24 port FXO/FXS cards digium sells. You could expand this as much as you want by adding more asterisk boxes with the 24 port FXO/FXS cards. Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 Conrad Wood wrote: Does anyone have any recommendations on how to connect 160 analogue phones to an asterisk PBX? Background information: A client wishes to replace their current PBX with a new VoIP system. Currently they have 2 PRIs. I intent to set up 2 asterisk PBXs with Debian GNU/Linux on raided drives. These drives will be mounted only read-only to recover gracefully from power-cycles. I am considering 2 ISDNGuards in front of the machines. More to the point: The client has 160 existing analogue telephones which they don't really want to change right now, because a) they are very cheap b) the users don't need to re-train. I have thought of Rhino Channelbanks, but then realised I need to use 7 of them and connect each with a T1. I don't really want to run 7 T1 + the 2 PRIs into one asterisk box for performance reasons. Ideally, several 48-Port SIP-FXS channelbank woulds be ideal I guess ;-). Does such thing exist? Or how do others do this? Conrad ___ --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] Two PBX
HELLO everyone I am having two alcatel 4600 digital phone PBXs .. They are situated in two locations 15km apart. I want users or extension in both PBXs to be able to dial and receive calls from each others through those 30 channels in the E1 .. I have line of sight so i am planing to use a wireless link between these two. Still i need a gateway or convertor from the PBXs to ip or lan ... Can i do this using two asterisk pc and two E1 card provided that the acatel has an E1 port in it . Is that possible to do this link ?? Can i make asterisk pcs transperant ?? What is the simplest configuration to make ??? Help regards to all Hazirliksiz yakalanmamak için MSN hava durumu hizmetinizde! Burayi tiklayin! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Fax Question
Michael Sampson wrote: Your best bet is to just not use fax machines. They are outdated technology. With email there is little reason to use fax machines anymore. But for some reason people just feel the need to hang on to them. There are still many valid uses for fax. The technology is not going away any time soon. While an electronic signature is legal in many cases, there are several where a signed document, even if it's faxed, is the only method that is allowed by law. The new technologies that allow scan to email are great (I prefer the scan to pdf instead of tiff), but they have their limitations. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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
[Asterisk-Users] Preferred editor(s) dialplan coding?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hey all, First of all, hello again! Been a while since I've posted to the list, but I've been here lurking and watching ;-) Anyway, I wanted to pose a general question to the list to see if it turns up new suggestions for everyone/me. What is your preferred editor when coding in the dialplan? This is mainly aimed at those of you who write the larger, more complex dialplans. I've been using UltraEdit, but would like to see if I can't find a better one, especially one with the ability to add-on and make it more Asterisk friendly. What I'm looking for: 1. Syntax Highlighting, and ease of updating that highlighting 2. Auto-updating lists (like sidebars) with: (this is a total WISH list) Variables Contexts a Command list? 3. SVN and/or CVS integration 4. Project ability 5. Macros (Macros are s handy!) 6. Autocompletion (and autocomplete edit ability) Anyway, just thought I'd put a bug in everyone's ear to think about. Cheers all! Sherwood -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) - GPGshell v3.50 iD8DBQFECKGnwWoA8HY7JXYRAra8AKDR8fdXmqHSw9sJSsTmnwEoeHOgzACfRu+Y BJiHMZZS+HIk6hRWLPRJOKo= =O2C5 -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] Echo Cancelation on TE110P
In theory I would say I agree how ever in practice... I have a PBX (Merlin Legend) that I am connected to via PRI (10 foot pre-fab'ed cable) and I get intermittent echo on the voip side. There is nothing in between * and the PBX... sean On Fri, 2006-03-03 at 13:42 -0600, Michael Sampson wrote: It is my understanding that when you hear echo the problem is on the other end. So if a caller complains they hear echo that is something you should be dealing with, but if you hear echo that is the phone companies fault. Now with a normal phone, the phone company will only echo cancel long distance calls. For local calls the latency is not high enough to matter. But with VOIP the added latency creates echo even for local calls. I think the reason you hear it on some numbers and not others is that the phone companies are doing echo cancel on some of those calls and not on others. Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 Kerry Garrison wrote: On a 55 station install onto a Cox PRI with a TE110P (Polycom 501 phones) a few users are complaiining about echo. According to the users, the echo seems to be phone number dependant. They claim that certain phone numbers have echo while others dont. Are there any tuning parametes like there is for a TDM400 card? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Fax Question
Michael Sampson wrote: Your best bet is to just not use fax machines. They are outdated technology. It is older technology, true... but certainly it's not useless technology. Certainly there is nothing yet to replace it properly. And I could argue this on a technological standpoint, and I have before on this list, but since you've such a closed-minded attitude towards older technology I don't think that there would be a point to it. E-mail is good and fun and has its uses. In some ways it has been able to replace fax communication, just like e-mail has been able to replace communications of other kinds as well. However, fax still has a purpose and a place, and many businesses still use it like crazy. With email there is little reason to use fax machines anymore. But for some reason people just feel the need to hang on to them. There still are many reasons to use fax, and yes, one of these is because so many people still have them and an analog phone line to boot. 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] snom 320 MWI light
I had the same problem, I just set the voicemail button on the phone to dial the voicemail extension, but you will still have the problem (at least on the 360) if the user uses the Soft buttons below the display to access the voicemail. On 3/3/06, Nabeel Jafferali [EMAIL PROTECTED] wrote: I am using a snom 320 running 5.3.6 with Asterisk 1.2.4. In the sip.conf entry, I have mailbox= [EMAIL PROTECTED] and vmexten=*98. The light on the snom 320 turns on when I have voicemail and the retrieve button dials the correct extensions. However, the light turns off immediately after making the call to voicemail, even if I do not check the voicemail.FYI Received the following from a vendor:Currently there is not a way to keep the MWI light to stay on after hittingretrieve button on the Snom.The best option at this point is to set checkmwi=1 in the general section of your sip.conf file.This will causethe light to turn back on shortly if there are un-checked messages waiting.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] CON-SNT-CP7970 resellers?
Anyone selling CON-SNT-CP7970 ? -Dan ___ --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] Echo Cancelation on TE110P
On Mar 3, 2006, at 11:42 AM, Michael Sampson wrote: It is my understanding that when you hear echo the problem is on the other end. So if a caller complains they hear echo that is something you should be dealing with, but if you hear echo that is the phone companies fault. Now with a normal phone, the phone company will only echo cancel long distance calls. For local calls the latency is not high enough to matter. But with VOIP the added latency creates echo even for local calls. I think the reason you hear it on some numbers and not others is that the phone companies are doing echo cancel on some of those calls and not on others. More likely some handsets are just louder, causing feedback through the far end mic. Sometimes you can fix this by reducing the gain on your mic. ie if your mic is too amplified, it's too loud at the other end, and is picked up by the far end mic and sent back to you as an echo. ___ --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] Spontaneous reloads
I am now receiving spontaneous restarts ofasterisk on my system, with no apparent rhyme or reason. I am using version 1.2.4, with zaptel-1.2.3 (downgraded from 1.2.4. I downgraded after the system started restarting spontaneously, this week. I upgraded last Friday). I see no indication of any problem in messages or full (which includes DEBUG messages). There are no problems being reported in dmesg or /var/log/messages. In other words, nothing is reporting to be in trouble. Is there another place that I can look for issues? Regards, Mark. This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system. ___ --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] Hardware Requirements for 1M minutes
On Mar 3, 2006, at 9:49 AM, David Thomas wrote: I'm doing an install for a client with the following requirements. - 1 Million minutes of outbound calling Per what? ___ --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] IAX2 register problem
On Mar 3, 2006, at 9:57 AM, Bartosz Jozwiak wrote: Hi guys, I am trying to register IP IAX2 phone to our Asterisk server. this is what I see on traffic debug between the asterisk server and IP phone. I do not see anything in asterisk console. Can somebody give me hints what could be the reason that phone is not registering? Thank you in advance. Bart Perhaps a firewall blocking 4569? 678.878478 62.204.64.161 -> 200.2.165.139 IAX2 IAX, source call# 2, timestamp 3ms ACK 680.576003 62.204.64.161 -> 200.2.165.139 IAX2 IAX, source call# 3, timestamp 20020ms PING 680.576241 62.204.64.161 -> 200.2.165.139 IAX2 IAX, source call# 3, timestamp 20023ms LAGRQ 681.652360 200.2.165.139 -> 62.204.64.161 IAX2 IAX, source call# 31736, timestamp 3ms REGREQ 681.653112 62.204.64.161 -> 200.2.165.139 IAX2 IAX, source call# 3, timestamp 3ms ACK 682.580584 62.204.64.161 -> 200.2.165.139 IAX2 IAX, source call# 3, timestamp 20020ms PING 682.580842 62.204.64.161 -> 200.2.165.139 IAX2 IAX, source call# 3, timestamp 20023ms LAGRQ 682.580920 62.204.64.161 -> 200.2.165.139 IAX2 IAX, source call# 3, timestamp 10024ms LAGRQ 682.581233 62.204.64.161 -> 200.2.165.139 IAX2 IAX, source call# 3, timestamp 17ms REGAUTH 682.874132 200.2.165.139 -> 62.204.64.161 IAX2 IAX, source call# 31738, timestamp 3ms REGREQ 682.874809 62.204.64.161 -> 200.2.165.139 IAX2 IAX, source call# 2, timestamp 3ms ACK snip>___ --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] a=fmtp:18 annexb=no
Hello Looking the SIP debug we see a change in the SETUP message from the Asterisk 1.0.x version to the 1.2.4. In the 1.2.4 we notice this line: a=fmtp:18 annexb=no This line cause problems in our plattform (We think our SIP - h323 gateway can't parse this line) Why this line its present in 1.2.4 version? Have anybody some clue? Regards JS. ___ --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] Setting Max Calls on an IAX trunk
Youll want to check the docco against the SetGroup and CheckGroup applications, although I think these have been deprecated in favour of a variable type approach now. Regards, Mark -Original Message- From: Marc Archer [mailto:[EMAIL PROTECTED] Sent: Friday, 3 March 2006 9:57 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Setting Max Calls on an IAX trunk Hi all, Is it possible to set the maximum number of simultaneous calls on an IAX trunk? I want to allow a maximum of 4 calls on our IAX trunk to a remote office, and then route any additional calls over the PSTN after that. I was thinking of keeping track of a count using the AstDB, but Im not sure how this would go if I had 4 calls incoming on the IAX trunk Anyone implemented something like this? Marc. ___ --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] Sipura RMA
On Mar 3, 2006, at 11:35 AM, Tom Vile wrote: I have had the same issue with Sipura as well, they gave me quite the run around on 1 bad 2002 ATA and decided to just forget about it and bought a new one to save time. So they gave you shitty support and you bought more? What are you a microsoft customer? ___ --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] new beta Grandstream firmware HT488_496_386
http://grandstream.com/BETATEST/HT488_496_386/ ___ --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
