Re: [asterisk-users] Which IP Phone is really the best?
That's like asking 'what is the best car in the world and why?' If you need to haul lots of people, you buy a minivan. If you live in the US and need to haul lots of people, you buy an SUV. If you need speed, to buy a motorcycle. If you need.. You need to decide what features are required, desired and how much you can spend on each phone. When I bought handsets, my primary requirement was that it had to be 100% remotely manageable and network booted/configd due to the auto provisioning software I wrote. I stuck to the Polycoms, Cisco, Linksys and Sipura brands. Within each brand will be several phones with distinct feature sets. Just figure out which one is required. My suggestion, find 4 phones you think will work and take them to the customer to choose. On Aug 31, 2007, at 1:11 PM, William Herrera wrote: I need to quote a client for a job and I was just wondering. Out of all the IP Phones out there, which one is the best and why? Thank you all, all opinions will be accepted. William Herrera LAN/WAN Technical Consultant ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which IP Phone is really the best?
Cisco 7960's: (SIPified) 1. Cheap 2. 6 lines is plenty 3. simple to config 4. stable On Jan 6, 2008, at 11:03 PM, William Herrera wrote: Alright, enough. At first I was to ignore to you all making statements like this one but I feel at this point that if I do not stop this it seems it will never stop. First thing first. I have a Bach. in Network Engineering. I did work for the Telefónica of Puerto Rico installing Asterisk (and working with Polycom, Cisco, Astra and Grandstream) for a bit over 2 years. I have been doing this now on my own business since October 2003 (www.lan-solutions.net), so I am not as you might think I am. I asked a simple question just to hear your opinion. It was not intended for so many of you waste your time (and mine) writing all this useless notes If you would have taken the same (or less) time just to answer the question (or to ignore it) we al would have been able to keep it simple, as intended... Case closed. WH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Cook Sent: Sunday, January 06, 2008 11:42 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Which IP Phone is really the best? Seriously, if you intend on proposing this to a customer it means you are selling your professional services. If you are asking questions like this, how successful do you expect your customer engagement to be? Even if someone recommends the best phone for your particular application, you will still have zero competency with it and spend inordinate amounts of learning time and re-work on the customer's time. Your inexperience will show. Customers are demanding and you will get thrown out on your a**. People expect IT to fail from time to time (unfortunately), but they expect 100% availability from their phones. Anything less and you will find yourself with a priority meeting at the client that includes your manager, CEO and their lawyer. Nothing travels faster than a bad reputation. Walk away. Research. Build a lab. Learn. - dbc. From: William Herrera [EMAIL PROTECTED] Subject: To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com I need to quote a client for a job and I was just wondering. Out of all the IP Phones out there, which one is the best and why? Thank you all, all opinions will be accepted. William Herrera LAN/WAN Technical Consultant ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 2767 (20080106) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com __ NOD32 2767 (20080106) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Job listing on cisco.com for Asterisk...?
I thought this was interesting, if you are in China and need a job, you might also... http://www.cisco.apply2jobs.com/index.cfm?fuseaction=mExternal.showJobRID=771671CurrentPage=1 * Working knowledge : Asterisk PBX; SIP Proxy Servers. ___ --Bandwidth 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] dial a pager and enter DTMF
I don't think you need the pipe in there. I've used this with the w option before, which adds a wait. Then continues .5 seconds later. RTM: http://www.voip-info.org/wiki-Asterisk+cmd+dial Try these: exten = s,2,Dial(SIP/TelaSip-gw4/5198881212D12345678) or exten = s,2,Dial(SIP/TelaSip-gw4/5198881212www12345678) also, you were missing a right parenth. -Original Message- From: [EMAIL PROTECTED] on behalf of Supa Sent: Sat 2/24/2007 9:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] dial a pager and enter DTMF Probably just a simple syntax issue, but does anyone know how to dial a number and the once phone has been answered, play DTMF tones and then disconnect. I am trying to use this for page notification. Ive been trying the following string with out luck: exten = s,2,Dial(SIP/TelaSip-gw4/5198881212|D(12345678) Any help would be greatly appreciated! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] cisco sip firmware update for cisco 7970
You can buy smartnet on a single phone for something like $8 a year. This will get you in legally. -Original Message- From: [EMAIL PROTECTED] on behalf of David Parcerisa Sent: Fri 2/23/2007 1:57 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] cisco sip firmware update for cisco 7970 I'm trying to buy the cisco firmware update but it seems that i cannot order online because I bought my 7970 on ebay. Is there any other chance to get this update? ... anyone can make me a favour and send it to me by email? 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 ___ --Bandwidth 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] SLA more than 100% ?
How does one answer more than 100% of the calls in less than 60 seconds? techsupp has 0 calls (max 20) in 'leastrecent' strategy (104s holdtime), W:0, C:3, A:2, SL:166.7% within 60s ___ --Bandwidth 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] addons 1.4 and cdr_addon_mysql not installed !
So, after reading this, I wonder if anyone has 1.4 and MySQL working... Is there a non-standard version I can download? more /usr/src/asterisk-1.4.0/doc/mysql.txt MYSQL LICENSING UPDATE == We were recently contacted by MySQL and informed that the MySQL client libraries are now under GPL license and not LGPL license as before. Since Asterisk does allow exceptions to GPL, we are removing MySQL support from standard Asterisk. We will, where appropriate, make it available via a separate package which will only be usable when Asterisk is used completely within GPL (i.e. not in conjunction with G.729, OpenH.323, etc). We apologize for the confusion. You may find this in the new asterisk-addons package. Mark Spencer Digium -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Friday, January 05, 2007 12:13 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] addons 1.4 and cdr_addon_mysql not installed ! On Fri, Jan 05, 2007 at 05:44:28PM +0100, Luca Lafranchi Lists wrote: Hi, I have installed asterisk on Ubuntu 6.06 server CD All required packages has been installed and upgraded When start sudo make menuselect As a rule, make as a user, make install as root. No need for sudo for anything other than 'make install' and such. from addons, I can't select all addons that require mysqlclient (app_addon_sql_mysql, cdr_addon_mysql, res_config_mysql). If I run apt-cache search mysqlclient, I find the following installed packages: libmysqlclient15-dev - mysql database development files libmysqlclient15off - mysql database client library You need the -dev one installed (recall that you're building a package. The relevant build dependencies according to the current Etch package: libmysqlclient15-dev asterisk-dev (It also requires libsqlite3-dev, but res_sqlite3 has a broken build process anyway and cannot use the system version of sqlite3) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] Guide to better performance using * ?
Can someone point me in the right direction to find documentation on best practices when setting up a new Asterisk server? I'm using RHES4 and Dell 1750 with TE412P. My current problems are frequent crashes and choppy audio so I think I can easily tweak these out of the picture. ___ --Bandwidth 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] Guide to better performance using * ?
It's a SCSI Raid, so not much: hdparm /dev/sda /dev/sda: HDIO_GET_MULTCOUNT failed: Invalid argument readonly = 0 (off) readahead= 256 (on) geometry = 8908/255/63, sectors = 73274490880, start = 0 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Withnall Sent: Wednesday, February 14, 2007 3:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Guide to better performance using * ? We had a similar issue with a 'new' server here. We had a trixbox install and the kernel didn't support the particular type of motherboard/drive combination and the disk was not in DMA mode. There was nothing we could do to get it to work and eventually put in an older motherboard. Since then, its been working beautifully. Run a hdparm /dev/hda (or whatever your disk is) and make sure its in dma mode. -- Kevin Withnall ILB Computing PH: 02 4227 0001 Mobile: 0412 453 846 FAX: 02 4253 0001 http://www.ilb.com.au/ http://kevin.withnall.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Thursday, 15 February 2007 5:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Guide to better performance using * ? Tim Connolly wrote: Can someone point me in the right direction to find documentation on best practices when setting up a new Asterisk server? I'm using RHES4 and Dell 1750 with TE412P. My current problems are frequent crashes and choppy audio so I think I can easily tweak these out of the picture. Ah! A Dell! What does your 'cat /proc/interrupts' say? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth 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] Guide to better performance using * ?
An older Dell that actually let's you steer everything else away from the Zaptel's IRQ... cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 0: 2768546973 0 0 0IO-APIC-edge timer 1: 27 0 0 0IO-APIC-edge i8042 8: 2 0 0 0IO-APIC-edge rtc 9: 0 0 0 0 IO-APIC-level acpi 11: 0 0 0 0 IO-APIC-level ohci_hcd 15: 20 0 0 0IO-APIC-edge ide1 201:2056566 0 0 0 IO-APIC-level megaraid 209: 2767719943 0 0 0 IO-APIC-level wct4xxp 225: 399626541 0 0 0 IO-APIC-level eth1 NMI: 0 0 0 0 LOC: 2768740418 2768740417 2768740416 2768740415 ERR: 0 MIS: 0 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Wednesday, February 14, 2007 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Guide to better performance using * ? Tim Connolly wrote: Can someone point me in the right direction to find documentation on best practices when setting up a new Asterisk server? I'm using RHES4 and Dell 1750 with TE412P. My current problems are frequent crashes and choppy audio so I think I can easily tweak these out of the picture. Ah! A Dell! What does your 'cat /proc/interrupts' say? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Rt db lookup
Okay. That doesn't help. What forces * to look at the DB rather than waiting on a registration ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Thomas Sent: Monday, January 15, 2007 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Rt db lookup On 1/15/07, Tim Connolly [EMAIL PROTECTED] wrote: Which command effects whether or not the * server will lookup a peer from the db even though the phone isn't registered locally? I have several * servers but I want any server to be able to lookup and send a call to phones registered on another server (SIP cluster?). You may want to look at DUNDi for this. http://www.dundi.info/ 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] OT: Quad-band cellphones with wifi stablesipsupport
Its not quad band and in my opinion doesn't perform well enough to be used for anything but basic email and phone calls. This phone, even on the newest version of firmware (Sprint) hangs when syncing with exchange to the point where you miss calls even though you tried to answer them. If you turn on wifi or Bluetooth, it simply compounds the problems. It will also require (literally) a ritualistic daily reset Notice the reset button on the bottom of the phone? Seriously, unless you live in an area where EVDO isn't offered even at 1x, forget the wifi. I don't think this phone has the muscle you want. My Treo 700wx outperforms my old PPC-6700 3 to 1 and doesn't lockup or need reset (rarely). The only really feature I lost was wifi and it took me two weeks to even notice this phone didn't have it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Norton - SophMedia LLC Sent: Monday, January 15, 2007 12:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Quad-band cellphones with wifi stablesipsupport Hey Tomer, I'm not sure if the Audiovox PPC6700 is quad band, but it does support Wifi and runnings SJPhone great! It is even usable over Sprints EVDO service. On Mon, 15 Jan 2007 08:01:44 +0200, Tomer Horn [EMAIL PROTECTED] wrote: Hello, I am looking to purchase a new quad-band cellphone and I'm looking for one with WiFi and enough CPU power for stable SIP calls. I was wondering if anyone here can share his experience and recommend on a good cellphone. Any chance there is such a phone with even good WiFi profiles management or am I asking for too much now? :-) Thanks, Tomer. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. ___ --Bandwidth 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] Rt db lookup
Which command effects whether or not the * server will lookup a peer from the db even though the phone isn't registered locally? I have several * servers but I want any server to be able to lookup and send a call to phones registered on another server (SIP cluster?). Thanks Tim ___ --Bandwidth 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] EM ?
When I send a call from my TE410P using EM, the legacy PBX answers the call but doesn't route it. Any idea what this could be? I assume the digits aren't being delivered properly to the legacy pbx. Any suggestions on what config settings to muck with? Asterisk SVN-branch-1.2-r40901 built by root @ pbx04 on a i686 running Linux on 2007-01-14 14:05:02 UTC zaptel.conf em=1-24 span=1,1,0,esf,b8zs,yellow em=49-72 span=3,3,0,esf,b8zs,yellow # Global data loadzone= us defaultzone = us zapata.conf [trunkgroups] spanmap = 1,1,1 spanmap = 3,2,3 [channels] switchtype=5ess signalling=em_w group=1 channel = 1-24 group=2 channel = 49-72 usecallerid=yes cidsignalling=dtmf Any suggestions? ___ --Bandwidth 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] Is there any Asterisk controllable thermostat?
My garage door is... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Crompton Sent: Monday, December 04, 2006 11:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is there any Asterisk controllable thermostat? I remembered I had an x10 bottlerocket in my X10 junkbox so I connected it to a spare serial port on my linux server (asterisk resides there) and implemented with some mods the code mentioned earlier http://lorance.freeshell.org/asterisk/#asterisk-can-control-the-world and it works great. Now I have one more way to control X10 devices. I can even call my VM on the way home and turn on my lights or whatever before I get home. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP rt load from db
Anyone know the command that tells * to load a sipfriend from the realtime db rather than saying no such host? I've tried various combinations of the rt commands: rtcachefriends=yes; ;rtcache=yes ;rtAutoClear=yes ;rtautoreg=yes ;rtIgnoreRegExpire=yes ;rtupdate=yes rtfromcontact=yes Basically I have a group of 4 * servers all routing calls, but only two are hearing the phones registration. I'd like the other two to load the sipfriends entry from mysql when a channel for that sipfriends is requested. Any ideas? ___ --Bandwidth 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: New Software available on Cisco.com P0S3-08-5-00
Fyi... My apologies if this is a dupe. -Original Message- From: Cisco Technical Support [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 13, 2006 8:52 AM To: Tim Connolly Subject: New Software available on Cisco.com New software images are available on Cisco.com for the product families that you have selected. If you would like to change your subscription, or unsubscribe, please see the bottom of this e-mail for instructions. This message serves only to advise of new patch availability on Cisco.com (http://www.cisco.com). This is not a direct recommendation to apply the described patch(es) to your system. Please use the release notes, readme(s), your sales team , your advanced services team, TAC, and above all your knowledge of your individual installation to decide if the patch is right for you. Newly Released Voice Software New Software at http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960 Filename: P0S3-08-5-00.zip Description : SIP Flash Image for 7940/7960 IP Phone v8.5(0) - Non-CallManager Filename: phrn85s.pdf Description : Release Notes for SIP Flash Image for 7940/7960 IP Phone v8.5(0) - Non-Call Manager Membership Maintenance: Please use these instructions to subscribe or unsubscribe from this list: 1. If you wish to subscribe or unsubscribe from all emails sent by Cisco, please visit your profile manager at http://tools.cisco.com/RPF/profile/profile_management.do to change your preferences. 2. If you wish to subscribe or unsubscribe from all/any software alerts and news, please visit http://www.cisco.com/cgi-bin/Software/Newsbuilder/Builder/VOICE.cgi to change your preferences. Cisco respects your privacy and is committed to protect the personal information that you share with us. Please review Cisco's policy statement at http://www.cisco.com/public/privacy.html, which describes how we collect and use your personal information. ___ --Bandwidth 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] Meetme monitoring (once)
Has anyone found a way to monitor a meetme conference for only the first user? I find have one recording per user is pretty hard on the server performance wise... Suggestions? ___ --Bandwidth 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] Meetme monitoring (once)
A little more RTFM'ing and voila! Using MeetMeCount I should be able to record only the first user. http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMeCount -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Connolly Sent: Tuesday, December 05, 2006 4:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Meetme monitoring (once) Has anyone found a way to monitor a meetme conference for only the first user? I find have one recording per user is pretty hard on the server performance wise... Suggestions? ___ --Bandwidth 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] Question about Realtime static table
This is more of a MySQL question.. But its going to look something like: ALTER TABLE `extensions_table` ADD `variable_name` type DEFAULT '0' NOT NULL ; From the specs page: http://dev.mysql.com/doc/refman/5.0/en/alter-table.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tielin Xu Sent: Tuesday, December 05, 2006 5:23 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Question about Realtime static table Hi All: I'd like to use Realtime Static in terms of the performance concern about dynamic realtime. Assume that I create a table: as following: CREATE TABLE `extensions_table` ( `id` int(11) NOT NULL auto_increment, `context` varchar(20) NOT NULL default '', `exten` varchar(20) NOT NULL default '', `priority` tinyint(4) NOT NULL default '0', `app` varchar(20) NOT NULL default '', `appdata` varchar(128) NOT NULL default '', PRIMARY KEY (`context`,`exten`,`priority`), KEY `id` (`id`) ) TYPE=MyISAM; Can anyone help me how to add variable names and values into the database? Thanks, Tielin ___ --Bandwidth 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] Monitor stops recording midstream?
Asterisk SVN-trunk-r7230 built by root @ pbx01.timsnet.com on a i686 running Linux on 2006-06-17 When I used monitor, I seem to get most calls cut off if they run very long. Sometimes two minutes, sometimes 5 or 15.. Seems random. Any ideas what might kill the recording process? I'm beginning to wonder if soxmix is truncating the file when it blends the in/outbound streams together due to bad data or something. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970 SIP won't update?
Does anyone know what triggers the 7970 to update its config? I was able to get it to update to SIP, but the config I used initially won't go away. I am making small changes to the SEPxxx.cnf.xml file and rebooting the phone, the phone is downloading the (TFTP) new config file, but I don't see any change on the phone itself. I've looked at the VersionStamp and incremented that, but still no go. Any suggestions? ___ --Bandwidth 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 in QUEUES dont work
I'm seeing the same issue, options tTH doesn't seem to help either... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rizwan HishamSent: Monday, July 17, 2006 3:25 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] DTMF in QUEUES dont work Hi,when im using only peer to peer call without any queues, im able to dial any extension or send any digit thru dtmf durng a call. but whenever i use queues then no phone dials any extension during a call or a conference. i cant even hangup a call using * key. Any ideas how this problem can be solved. im using H323 and SIP channels and i have set both channels to use dtmf=rfc2033.-- RegardsRizwan HishamSoftware Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7960 SIP 8-3-0 getting Got SIP response 400
After upgrading my phones I now see routine error messages: -- Got SIP response 400 Bad Request back from 10.5.1.94 Asterisk SVN-trunk-r7230 Cisco 7960 SIP version 8-3-0. Sip show peer: * Name : 14012 Secret : Set MD5Secret: Not set Context : labcm33 Subscr.Cont. : Not set Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : 14012 VM Extension : asterisk LastMsgsSent : 0 Call limit : 0 Dynamic : Yes Callerid : removed Expire : 272931 Insecure : no Nat : Always ACL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 10.5.1.94 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 14012 SIP Options : (none) Codecs : 0x8000e (gsm|ulaw|alaw|h263) Codec Order : (ulaw,alaw,gsm) Status : Unmonitored Useragent: Cisco-CP7960G/8.0 Reg. Contact : sip:[EMAIL PROTECTED]:5060;transport=udp Any ideas? The phones seem to work fine other than the annoying console message. Is there some secret setting I can add to my config to stop this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7960 SIP 8-3-0
Looks like the MWI broke on 8-3 also... ___ --Bandwidth 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] Using HINT with Cisco 7960/SIP
Can someone provide an example of how to use HINT priority with Cisco 7960/SIP phones? I don't fully understand what exactly the hint does, but I believe it mimics a legacy PBX's bridge-appearance function. Is this correct? ___ --Bandwidth 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 version of NVBackgroundDetect:
Justin asked me to post a note about a new version of NVBackgroundDetect coming out very shortly. Be patient! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do I limit the lenght of a call
Google is your friend, or you enemy, either way they usually have an answer: /Dial a single destination, ringing for a maximum of 20 seconds. Limit the call length to 60 seconds, warning the caller when only 20 seconds remain:/ exten = 200,1,Dial(SIP/1234,20,L(6:2)) -- http://www.automated.it/asterisk/lah-3-6-05_4.html John Rich wrote: Hi, Is there a way to limit the duration of a call in the Dial command? Mainly for perpay account. Thanks __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 6.3 unlock/reset?
I've had a few, even on 7.4+, that were impossible to recover the password. I usually end up looking at the current network settings and putting an IP alias on my tftp server so it will answer the tftp get requests coming from the phone. It gets tricky when the original config has the TFTP server and the phone on seperate subnets. Guess this is bad karma for buying used gear... I can deal with that if it saves me $300/phone! Shaun wrote: Anybody know the proceedure to factory reset the a 7960 phone running 6.3 SIP software? I've tried holding # when booting the phone and nothing, i can do that on my 8.2 phone but this phone i just got with 6.3 isnt working. Also **# doesnt work either.. ___ --Bandwidth 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] Performance: Xeon or Opteron?
I was offered an upgrade path for my two Dell 1750's (2.8 Dual Xeon) to get into a pair of new Dual Core Dual Opteron servers. Assuming I can get the IRQ BS worked out so my TE411XP doesn't flip out, this should be a pretty significant upgrade. Has anyone been able to quantify any benefits to using one processor over the other? Should I wait for the newer Intel processors this summer or go for the AMD DC DO? Thanks Tim ___ --Bandwidth 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] res_config_mysql.so: undefined symbol: __stack_chk_fail
Did you upgrade all the mysql packages, or just the server? I would bet you missed the -dev or -lib package. kritikus Araklidas wrote: Hi everyone: I installed the lates version of Asterisk with Asterisk Add-Ons. A month ago i upgraded my database form mysql 4.1 to mysql 5.0. So after to start Asterisk i have the following error: [res_config_mysql.so]Apr 11 17:25:51 WARNING[31300]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: __stack_chk_fail Apr 11 17:25:51 WARNING[31300]: loader.c:554 load_modules: Loading module res_config_mysql.so failed! Any idea.?? Regards. Cristian. ___ --Bandwidth 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 to MySQL Data Lookup Warning Message?
I've been seeing this for a while. No clue how to fix. The source I have
from my last update says extra_log=0, so it shouldn't be showing this
message at all...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JR
Richardson
Sent: Tuesday, March 28, 2006 4:22 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk to MySQL Data Lookup Warning Message?
Hi All,
I'm getting a strange warning message when I perform a MYSQL data
lookup. The operation performs fine, I retrive the data I'm looking for
and continue on through the dial sequence without an issue. I'm
wondering if this warning message is something to be concerned about,
can't find any info about it.
warning message:
Mar 28 15:55:40 WARNING[27481]: app_addon_sql_mysql.c:318 aMYSQL_fetch:
ast_MYSQL_fetch: numFields=1
Should I be concerned? Does anyone know what it means?
Thanks.
JR
I'm dialing extension 1001
extensions.conf
[lookupmysql]
exten = _X.,1,MYSQL(Connect connid 10.10.10.110 asteriskdb password
table) exten = _X.,2,MYSQL(Query resultid ${connid} SELECT\
fullcontact\ from\ table_sip\ where\ name=${EXTEN}) exten =
_X.,3,MYSQL(Fetch fetchid ${resultid} var1) exten = _X.,4,MYSQL(Clear
${resultid}) exten = _X.,5,MYSQL(Disconnect ${connid}) exten =
_X.,6,GotoIf($[${var1} = ]?invalid,i,1:${EXTEN},7) exten =
_X.,7,ChanIsAvail(SIP/${EXTEN}IAX2/${EXTEN}|sj)
exten = _X.,8,Set(direct=${var1:4})
exten = _X.,9,Dial(SIP/${direct},30,r)
exten = _X.,10,Hangup
exten = _X.,108,Goto(sendtovm,${EXTEN},1) exten = _X.,109,Hangup
asterisk-cli
-- Goto (lookupmysql,1001,1)
-- Executing MYSQL(SIP/1239-fc6c, Connect connid 10.10.10.110
asteriskdb password table) in new stack
-- Executing MYSQL(SIP/1239-fc6c, Query resultid 1 SELECT
fullcontact from complete_sip where name=1001) in new stack
-- Executing MYSQL(SIP/1239-fc6c, Fetch fetchid 2 var1) in new
stack Mar 28 15:55:40 WARNING[27481]: app_addon_sql_mysql.c:318
aMYSQL_fetch: ast_MYSQL_fetch: numFields=1
-- Executing MYSQL(SIP/1239-fc6c, Clear 2) in new stack
-- Executing MYSQL(SIP/1239-fc6c, Disconnect 1) in new stack
-- Executing GotoIf(SIP/1239-fc6c, 0?invalid|i|1:1001|7) in new
stack
-- Goto (lookupmysql,1001,7)
-- Executing ChanIsAvail(SIP/1239-fc6c, SIP/1001IAX2/1001|sj)
in new stack
-- Executing Goto(SIP/1239-fc6c, sendtovm|1001|1) in new stack
-- Goto (sendtovm,1001,1)
JR Richardson
Engineering for the Masses
___
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[Asterisk-Users] TAC Case Cisco 7960 Proxy address showing up in callerID
Figured this was worth passing on... This was reported due to the proxy IP address showing up in CallerID on the phone. -Original Message- Sent: Thursday, March 23, 2006 12:01 PM Tim, I have tracked down the source of the change in the SIP firmware. The behavior was changed as a fix to bug id CSCsc22406 (host part of the callerid not preserved in ReceivedCall entry). This was a problem when calls came from an outside domain since return calls would not be sent to the right server. The fix was to always preserve the domain or ip address from the From field. There is currently a discussion going on about whether this is the correct behavior. I will let you know what I hear as the situation develops. 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] Dialplan : Forwarding call to voicemail after onering iif extension is busy
Sure, just make your voicemail wait 5 seconds before answering the call. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Navneet ShahSent: Thursday, March 16, 2006 10:45 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Dialplan : Forwarding call to voicemail after onering iif extension is busy Hello. Is there any way to forward incoming call to voicemail in one ring if the person on the extension is busy. Regards --- Navneet Shah Systems Administrator YL Consulting, Inc. 210-340-0098 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?
I'm not sure this is the issue. Every call seem to get the proxy address added whether it's the main proxy or the backup. What has to match to make the phone NOT append the proxy address? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Connolly Sent: Wednesday, March 15, 2006 1:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy? That's probably what is happening on my end. Any suggestions on how to fix this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Tuesday, March 14, 2006 7:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy? We only had the problem when the call was redirected from one server to another. So if a phone was called from another phone on the server, the called worked perfectly, but if it was redirected from another server, we got the proxy added to the end. Doesn't help when you're trying to make the existence of multiple servers transparent. Aaron Chris Stenton wrote: Maybe I have something strange in my dial plan but I have no problem just hitting dial from missed calls under 8.2. Chris - Original Message - From: Aaron Daniel [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 13, 2006 8:44 PM Subject: Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy? We rolled back to 7.4 cause of that too. 7.5 has a strange bug where if the server loses connection, the phone's just don't try re-registering. Aaron Tim Connolly wrote: Just curious, why not 7.5 ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Monday, March 13, 2006 2:28 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy? I'm using P0S3-08-2-00.. I noticed the callerID started showing up with the number, then @proxy-addr... So the callerID on the phone looks like: [EMAIL PROTECTED] which of course is logged in the missed calls exactly like that, and completely foobars the dialing string if you try to dial a missed call by simply hitting the dial button. Can anyone else verify this problem? Yeah, that bothered me so I rolled back to SIP 7.4. 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?
That's probably what is happening on my end. Any suggestions on how to fix this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Tuesday, March 14, 2006 7:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy? We only had the problem when the call was redirected from one server to another. So if a phone was called from another phone on the server, the called worked perfectly, but if it was redirected from another server, we got the proxy added to the end. Doesn't help when you're trying to make the existence of multiple servers transparent. Aaron Chris Stenton wrote: Maybe I have something strange in my dial plan but I have no problem just hitting dial from missed calls under 8.2. Chris - Original Message - From: Aaron Daniel [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 13, 2006 8:44 PM Subject: Re: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy? We rolled back to 7.4 cause of that too. 7.5 has a strange bug where if the server loses connection, the phone's just don't try re-registering. Aaron Tim Connolly wrote: Just curious, why not 7.5 ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Monday, March 13, 2006 2:28 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy? I'm using P0S3-08-2-00.. I noticed the callerID started showing up with the number, then @proxy-addr... So the callerID on the phone looks like: [EMAIL PROTECTED] which of course is logged in the missed calls exactly like that, and completely foobars the dialing string if you try to dial a missed call by simply hitting the dial button. Can anyone else verify this problem? Yeah, that bothered me so I rolled back to SIP 7.4. 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?
Just curious, why not 7.5 ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Monday, March 13, 2006 2:28 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Cisco 7960 8.2 callerID lists proxy? I'm using P0S3-08-2-00.. I noticed the callerID started showing up with the number, then @proxy-addr... So the callerID on the phone looks like: [EMAIL PROTECTED] which of course is logged in the missed calls exactly like that, and completely foobars the dialing string if you try to dial a missed call by simply hitting the dial button. Can anyone else verify this problem? Yeah, that bothered me so I rolled back to SIP 7.4. 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 8.2 callerID lists proxy?
I'm using P0S3-08-2-00.. I noticed the callerID started showing up with the number, then @proxy-addr... So the callerID on the phone looks like: [EMAIL PROTECTED] which of course is logged in the missed calls exactly like that, and completely foobars the dialing string if you try to dial a missed call by simply hitting the dial button. Can anyone else verify this problem? 7.5 and 7.4 don't do this. I just rolled back and 7.5 works as expected... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom 4000 results?
Has anyone tried the Polycom 4000 on SIP/* ? ___ --Bandwidth 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: Ring requested on channel already in use - fix
I'm replying to this mainly to add my comments to the archive and then all the webcrawlers... I found a deprecated command curl which I though had simply been converted from an app to a function, was actually completely non-working. Anytime my call hit a exten = s,1,set(CURL=curl()), the channel would get hung up. Almost immediately, the call would retry on the same channel and get the message Ring requested on channel I'm not sure if it was because it was being called pre-answer or if some portion of the curl function still exists, but either way, it totally disabled our inbound calls as each and every call used that curl function to replace the callerIDname variable. The fix was simply to remove all mentions of curl. Hope this helps someone else... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of alan Sent: Monday, September 26, 2005 1:28 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Ring requested on channel already in use I posted this 1.2.0-beta1 success story to asterisk-dev, and someone recommended that asterisk-users might benefit from it as well. Thanks, Alan Ferrency pair Networks, Inc. [EMAIL PROTECTED] -- Forwarded message -- Date: Thu, 22 Sep 2005 17:35:08 -0400 (EDT) Subject: [Asterisk-Dev] Re: Ring requested on channel already in use To: asterisk-dev@lists.digium.com alan wrote: A problem was recently posted on the Asterisk-Users mailing list, and it went unresolved. Now that it's plaguing our production system as well, I need to look into it further. Good report, lots of information. See if you can reproduce it in CVS-HEAD (Asterisk, libpri, zaptel) snip You need to test this with cvs head (1.2beta) first to see if it's not already fixed... I am happy to say that since we upgraded to 1.2.0-beta1, our problems with Asterisk instability have not recurred. Our uptime is over a week, with the last restart a result of the upgrade. Thanks! I didn't like to see the answer upgrade your production system to a beta version, but the truth is, it was working poorly enough that it was basically impossible not to at least try it. Here is a summary of the symptoms we were seeing in 1.0.9, for others with this issue who may benefit from an upgrade: We narrowed the problem down to this sequence of events: - an incoming Zap call on a PRI channel - was sent to the queue - and answered by a AgentCallbackLogin queue agent - who was using a SIP phone - and the agent attempted to SIP REFER transfer the call - to another AgentCallbackLogin agent on a SIP phone That's a lot of channels (zap - agent - local - sip, transferring to agent - local - sip). When this happened, we saw these symptoms: - Rarely, the transfer succeeded. - More often, the ZAP channel was put in limbo and both SIP parties were dropped; or the transfer completed but there was one-way audio from Zap to SIP only. - Often, when the transfer failed, Asterisk was left in an inconsistent state, and would not function correctly until a restart was performed. -- asterisk -r consoles could not execute commands successfully -- sip show channels produced bogus output -- incoming Zap calls (over a PRI) resulted in Ring requested on channel... already in use errors, and the calling party was dropped immediately. After this experience with 1.2, I'd say that the upgrade should not cause many problems, as long as you thoroughly research and implement all required configuration changes. We have not experienced any problems with 1.2 which weren't also problems in 1.0.8/9, but we have had many other little issues solved which we were previously trying to ignore. Thank you very much, Alan Ferrency pair Networks, Inc. [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MOH broke with 1.2.4 .. ?
/etc/asterisk/musiconhold.conf: [default] mode=files directory=/var/lib/asterisk/mohmp3 application=/usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s -- Executing Answer(Zap/1-1, ) in new stack -- Executing MusicOnHold(Zap/1-1, ) in new stack -- Started music on hold, class 'default', on channel 'Zap/1-1' -- Stopped music on hold on Zap/1-1 I've got three mp3 files that worked fine on the latest cvs-head version. With the upgrade to 1.2.4, I get no audio whatsoever. Any suggestions? I cranked up verbose to 255 with no extra info.. Same with debug. ___ --Bandwidth 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: [Aterisk-Users] Zapbarge feature available?
Were you able to acomplish this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan Sent: Thursday, October 27, 2005 5:31 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Zapbarge feature available? We would like to beable to listen in and interact with the person in a queue, talk to our agent and NOT have the other person be able to hear us. Is there a way to do this? Kyle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation 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] TE405p -- loopback for the phone company?
I wonder if Digium has any intentions of fixing this. I brought this to their attention shortly after purchasing a pair of TE411's. You can issue a loopup on span 2 only to get a message saying looping span1 which is to say, a bit scary when you only have two active PRI and one is already down for testing... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Wednesday, January 11, 2006 9:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TE405p -- loopback for the phone company? Eric Lyons wrote: I got zttool running and selected loop on the interface, but it didn't seem to do what they wanted (nor could I tell that it did anything at all). Many googles for zaptel and loop didn't turn up anything useful. This is a bug that needs to be fixed; currently the dual-/quad-span drivers to not respond to remote loop-up requests, nor do they have any mode to loop data back towards the network. ___ --Bandwidth 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] TE405p -- loopback for the phone company?
Hmm.. I'm running CVS-head from a few days ago. TimeWarner said they couldn't loop me, so I plugged in a router and they were able to loop and test the PRI. Is there any way to do loops from within the Asterisk console? I typically use zttool. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Monday, February 06, 2006 4:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TE405p -- loopback for the phone company? Tim Connolly wrote: I wonder if Digium has any intentions of fixing this. I brought this to their attention shortly after purchasing a pair of TE411's. You can issue a loopup on span 2 only to get a message saying looping span1 which is to say, a bit scary when you only have two active PRI and one is already down for testing... It has been fixed in Zaptel 1.2.3. The dual/quad span cards now properly respond to loop-up/loop-down requests initiated from the other end of the span. ___ --Bandwidth 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] Realtime Queues and Agents
Wouldn't it be easier to keep the agents in the table all the time, and simply update the logged_in status column for that row? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Tuesday, August 30, 2005 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime Queues and Agents Julian Lyndon-Smith wrote: We use agents and queues, with CVS HEAD. I've read up on realtime queues and queue members, (and actually understand it!) but there is no reference to agents. Is it possible to have realtime agents as well ? Julian. No there isn't. And there won't be until RealTime gets updated with 'INSERT' and DELETE abilities. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation 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 sip firends not being updated
I've got realtime sipfriends running pretty well. One this I noticed is that if I make a change to the DB, the server's 'sip show peer 1234' never shows the update until after I do a 'sip reload'. My info, cvs-head from 12/17/05on a Dell 1750. the mysql db is on a seperate server, as so is the voicemail app. Any idea how often rtcachefriends=yes will allow the server to reverify the SIP friends entry for an 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] MWI not working - using seperate vm and call routers:
I have a pair of servers tied to PRI's which only do SIP/ZAP/IAX terminations and routing. The IVR/Voicemail is on a separate server which accepts calls from the front line servers via Iax. I am trying to use the MWI on our Cisco 7960 phones, which isn't working, but I think its because the voicemail server doesn't have the phones in its list of sip peers. Does anyone know of a way to fix this? I tried loading my sipfriends realtime table on the voicemail server, but since the phones don't ever register which that particular server, the voicemail server's 'sip show peers' is always empty. Any suggestions on how to get MWI working? Is there someway for the voicemail server to tell the front line server Hey, 1234 has voicemail, light him up! ? Thanks Tim ___ --Bandwidth 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] MWI not working - using seperate vm and callrouters:
Thanks for the suggestion. That sounds very easy and not much code involved. Wow.. I just tested that by touching msg.txt and it works great! Thanks again, Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Wednesday, December 21, 2005 1:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MWI not working - using seperate vm and callrouters: There are a few methods of doing this without rewriting code... On our system, we have 2 asterisk servers that actually run the phones, a voicemail server, and a gateway... whenever a call comes into the voicemail server, we have an 'h' extension that checks to see if the person does in fact have voicemail, and then touches or removes a file in the phone's mailbox on the two primaries, which cause the MWI light to come on. It's probably considered half-assed, but it works like a charm for us. Aaron Olle E Johansson wrote: Tim Connolly wrote: I have a pair of servers tied to PRI's which only do SIP/ZAP/IAX terminations and routing. The IVR/Voicemail is on a separate server which accepts calls from the front line servers via Iax. I am trying to use the MWI on our Cisco 7960 phones, which isn't working, but I think its because the voicemail server doesn't have the phones in its list of sip peers. Does anyone know of a way to fix this? There is no communication between your SIP registration server and the voicemail Asterisk. The only way to fix this is to start developing new code :-) /O ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New voicemail alert options for Cisco 7960 SIP phones
I'm looking for ideas on how to implement voicemail notification on Cisco 7960 SIP phones. Something like a light on the legacy pbx-phone would be perfect. Even maybe go so far as a quick ring to the extension every 15 minutes or so, but then that would increment the on-screen missed call count. How about a debug-test call where we telnet into the users phones, open a test call on speakerphone back to some extension which simply plays a soundfile like You've got mail. Any suggestions? Is there any simple way to check the voicemail application to see which mailboxes have new messages waiting? Is there simple way to notify users on phones like the Ciscos ? Thanks Tim ___ --Bandwidth 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] Backend network, one-way audio...trunking
I've got a cvs-head box running RHES4. I have it spread across two vlans as I am using 802.1q trunking to my cisco switch. On the front side, everything works great. If I move a phone from the front to the rear, change its IP address and its config to reflect that move, and update the SIP config, I end up with one-way audio. This server also runs vrrp, but I'm using the real interface (vlan interface) as the destination of the sip. I'll post a sip debug tomorrow if we can't figure this out... Thanks Tim ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Odd problem with sip.conf register command:
Asterisk cvs-head (up to date) keeps core dumping on me. I finally tracked it down to my register command for Vonage in the sip.conf file. If I remove the username and password from the register command, it won't core dump, but of course won't register either... This is odd. Any suggestions? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF being cancelled
Ive got an application where I need to simply dial the console (local sound card using OSS driver) and pass any DTMF tones to the console. No matter whether I come in on a zap/sip/iax channel, the DTMF is always being muted. Is there anyway to disable this? Im not specifying any Dial options, so it shouldnt be the tT issue Thanks Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Plantronics USB Headsets Audio 45
SJPhone and a few others. Seems to work well. A little small for my head insert joke here though. Not bad for $50. The Logitech is right up there with it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Tuesday, August 16, 2005 12:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Plantronics USB Headsets Audio 45 Anybody using Plantronics USB headsets? What softphone are you using and whats your overall experience? Any comments/suggestions? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Load Testing
Almost.. A call on hold doesn't represent the true bandwidth and CPU that a *real* call utilitizes. Short of producing an echo or feedback on each call to make it look like a real call, I'm not sure how you could create a real call test scenario. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of The VoIP Connection Sent: Friday, August 12, 2005 10:42 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Load Testing Anton, A great tool for ghetto call capacity testing is a single snom phone. There is no limit to how many calls a snom phone can make, just put it on hold and dial again. So, with a single snom phone and a little imagination you can test any number of scenarios. You can approximate basic SIP capacity by creating an extension that plays the asterisk test message and dialing it repeatedly until quality starts to degrade or asterisk gives up. To simulate actual call throughput you really need another (faster) machine to connect to, but you can use the same technique. You can run top on the console while you are doing your tests to see what resources you are using. Check your logs when you are done to see what errors were generated when it came unglued. CPU is not always the limiting resource, especially with Digium card interfaces which tend to be bound by FSB speed, but echo cancellation and codec conversion will burn a LOT of cycles. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Anton Krall [mailto:[EMAIL PROTECTED] Sent: Friday, August 12, 2005 9:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Load Testing Guys. How and which tools to use to load test an asterisk install? Say for example, you need to see how many calls can be routed thru before losing quality and making the cpu jump to the roof? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Suggestions for mainstream hardware compatiblewith TE411P.
Thanks for the suggestion. One of my problems is that a TE110P worked flawlessly in my MPC server. As soon as I upgraded to the TE411P, I started having all sorts of issues. The biggest being an IRQ conflict, which was resolved but only to find I still get kernel panics under minor load. I think Im finding myself victim of early-adopter syndrome. I havent been able to get much feedback (no pun intended) from owners of the TE411Ps. Anyone want to trade a TE411P for 4 TE110Ps ? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Saturday, August 13, 2005 12:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Suggestions for mainstream hardware compatiblewith TE411P. On Aug 12, 2005, at 7:06 PM, Tim Connolly wrote: I checked the list of what not to use, but am still having no fun trying to find a working box. Can someone suggest a Compaq or Dell or MPC or any other brand and model that is known to work well with the TE411P ? Will an old Proliant do? I've built five PBXes on Dell Dimension 2600s that run flawlessly. They're P3 2.6GHz machines, so processor load stays super-low. Using a combination of TE110Ps and VoIP termination/origination, across ~35 users at each location on 7960s. Never missed a beat. I would consider a consumer box with a strong CPU over an old server, then spend your money on an ATA RAID card and mirror everything for disaster recovery. Hope that helps. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best Voip provider (Broadvoice and Vonage comparison)
If you need a FXS, Vonage starts at $15. If you want to simply go soft-only, Broadvoice would be a better choice. After the marketing and all the features that nobody uses are thrown out, it comes down to consistency. Broadvoice has had some problems in the past 6 months, Vonage hasn't (that I know of). Vonage makes you throw away your ATA and won't let you reactivate it if you close your account. Broadvoice has free calling to US + 34 countries for a $25 account. I can setup a broadvoice account and have it up and running as quickly as I can click through the order process, login to get my password, and then setup the SIP config on my asterisk box. Vonage...not so much as you have to either activate a brand new retail unit or allow them to ship you one. Granted, you can add soft accounts once you have a hard account, but soft phone accounts are stuck at 500 free minutes. Choose your poison... No matter who you go with, there will be pros and cons. I guess by default, I choose Broadvoice, only because Vonage makes it hard to purchase only what I need. There are others out there too, cheaper, better, blah... Its not worth arguing over. They all have outages eventually, piss of a few active mailing-list'ers and suddenly their reputation goes to downhill. Vonage - if you are reading this, stop requiring people to buy a device just to get a softphone activated. Also, stop making us buy a NEW device if we decide to disconnect the old device for a while. While your at it, put an unlimited minutes option on your softphones. Look at Broadvoice's plans! Broadvoice - if you are reading this, please let me register with all your proxies, not just one! Also, stop answering my call just to play some corny message that says my call won't go through. I can reroute the call if you just congest the call invitation. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jonny hashem Sent: Saturday, August 13, 2005 2:43 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Best Voip provider what is the best voip provider that provides good service ,good voice quality and good rates . any one have an experience with voip providers advice me. Regards; jonny Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Firewall will definately increasejitters inyourvoice conversation
On that note... IPSec tunnels seem to reek havoc on the echo canceling/training process. Anytime our Cisco PIX loads up, the echo complaints start coming in. Stay away from the IPSec tunnels. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Travers Sent: Saturday, August 13, 2005 5:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Firewall will definately increasejitters inyourvoice conversation Rich Adamson wrote: That's a crack of crap sold by the marketing (not sales) people selling firewalls. If you know what you're doing, one can very easily secure any linux system to function on the Internet (etc) without a firewall. It all depends on your level of knowledge/skills on how to disable those items that are not really needed in your environment. Start with a 'netstat -a' to identify those ports that are listening, and shut those items down that you don't want exposed. You can do the same for any MS system as well. But you still want a firewall here especially if you have several VOIP systems which could be making independent connections to the internet. The firewall in this case will hopefully not only do things like VPN for securing your data in trasit between your office and a remote one, but it will also provide a platform for QoS/traffic shaping. To avoid the firewall here is actually *asking* for sound quality problems in addition to the fact that you no longer have the entrence point to your network secured. Now to your point Almost any Linux system can be configured (if you know what you are doing) to perform all these firewalling functions. Just add an extra network card, put it on the perimeter of your network, set up iptables, traffic shaping, uninstall unnecessary software, use Netstat to doublecheck listening ports, etc. and you have your firewall. A firewall doesn't have to be expensive but some form of perimiter control is very helpful in these cases. Best Wishes, Chris Travers Metatron Technology Consulting ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] vmail.cgi
You might try to su - apache and make sure apache can read the file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Vega Sent: Saturday, August 13, 2005 5:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] vmail.cgi I'm trying to get the vmail.cgi script to work. Followed the instructions in the wiki, but I'm getting stuck with this error: Bleh, no /etc/asterisk/voicemail.conf at /var/www/cgi-bin/vmail.cgi line 96. I chmodded the files and directories used by vmail.cgi per the wiki instructions, but it appears Apache can't access anything oustide /var/www I'm running CentOS4/Apache ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Load Testing
I could probably shoot about 115 calls towards you, would that do ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Friday, August 12, 2005 8:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Load Testing Guys. How and which tools to use to load test an asterisk install? Say for example, you need to see how many calls can be routed thru before losing quality and making the cpu jump to the roof? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE411P problem
You might start by running /usr/src/zaptel/zttest. See if you stay at 100%. That's going to be the first thing digium checks. You might also run the autosupport script and take a look at it for anything obvious. I'm having lots of stability problems with my 411's. I'm not blaming the 411 yet, just seems odd that I ran for months on a TE110P with a peak of 10-15 calls, and now my box kernel panics each time it hits the same load. Granted, its got 4 PRI's now, but still only 10-15 calls will kill it. Does seem to kill the echo as long as the zttest comes back clean. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Rees Sent: Friday, August 05, 2005 6:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TE411P problem List, I just tried to swap out our 410 for a 411 and we started have problems with on of our T1's. Setup: Span 1 - Dedicated PRI for long distance. Span 2 - 12 channels fxs_gs outgoing local. 12 Channels em_w incoming DID's. I didn't have any problems with the PRI. The trouble was with the T1. We were unable to place any local calls, and all incoming DID's where garbled. What I mean by garbled, 7744 would come in a 44. I turned off all of the software echo cancel stuff in the zapata.conf. I am going to email Digium on Monday, but I am fishing here. This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Suggestions for mainstream hardware compatible with TE411P.
I checked the list of what not to use, but am still having no fun trying to find a working box. Can someone suggest a Compaq or Dell or MPC or any other brand and model that is known to work well with the TE411P ? Will an old Proliant do? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] z-machine + asterisk = fun!
Wow! Not sure what else to say. This ranks right up there with my ability to open my garage door from asterisk... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, August 07, 2005 1:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] z-machine + asterisk = fun! I was tinkering with Asterisk and the Festival text-to-speech engine, and wrote some short Asterisk::AGI scripts to read back live weather reports. After that, I thought I needed something more interactive to work with... Then I had a flashback to 1996, first year university, standing in the C O club at the University of Waterloo, where someone had just pulled out their US Robotics Palm Pilot and started up Zork. A couple of hours later, after a quick trip to the campus computer store, I was playing Zork in the palm of my hand! Now Zork is back! Listen as the eerie voice of Festival takes you into the Underground Empire, and marvel as you explore this world with your dial pad, unlocking the secrets within! Note that some more commands need to be implemented before you can actually -enter- the underground empire. For now you can just futz around on the surface. See $dtmf_translation in Asterisk/Games/Zork/ZIO_Asterisk.pm for number-to-phrase translations. I've posted the proof-of-concept at http://uc.org/read/Zasterisk Feedback is welcomed ;-) Cheers, Simon P. Ditner | The Toronto Asterisk Users Group -- http://taug.ca | Join by sending email to [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Native Bridge killing audio, sending dtmf
I'm seeing this same issue. The following message will popup on the console: -- Attempting native bridge of Zap/1-1 and Zap/74-1 At the same time my call is briefly muted, I hear a quick DTMF tone, then it unmutes. The whole process takes about 1.5 seconds. Is there any way to stop these attempts? My call was an inbound PRI to zap te411p call, back out another span on the same card to a legacy pbx. Any suggestions? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Wednesday, June 22, 2005 11:33 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Native Bridge Guys., How can I disable native briding on sip? I get this but after that, the call just tries to do the bridge and freezes == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing Dial(SIP/demo-3763, SIP/demo2|20|mwtWT) in new stack -- Called demo2 -- Started music on hold, class 'default', on SIP/demo-3763 Jun 22 23:31:46 WARNING[31090]: chan_sip.c:2901 find_call: Call missing call ID from '201.129.249.85' -- SIP/demo2-07af is ringing -- SIP/demo2-07af answered SIP/demo-3763 -- Stopped music on hold on SIP/demo-3763 -- Attempting native bridge of SIP/demo-3763 and SIP/demo2-07af == Spawn extension (telefonos, 102, 1) exited non-zero on 'SIP/demo-3763' Both phones are inside a NATted network connecting to a remote asterisk, the eco test works so audio works great on each phone to asterisk. My sip.conf file has canreinvite=no and nat=yes on both phones config. Any ideas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 64K ISDN call not passing thru
I'm trying to pass a 65K DATA call in one channel on my Digium TE411P to another channel on a different span. Any idea what could keep this call from going through? -- Accepting call from '' to '5444' on channel 0/1, span 1 -- Executing Goto(Zap/1-1, sendto-definity|5444|1) in new stack -- Goto (sendto-definity,5444,1) -- Executing Dial(Zap/1-1, ZAP/g2/5444) in new stack -- Requested transfer capability: 0x08 - DIGITAL -- Called g2/5444 -- Zap/49-1 is proceeding passing it to Zap/1-1 -- Channel 0/1, span 3 got hangup request -- Hungup 'Zap/49-1' == No one is available to answer at this time (1:0/0/0) -- Channel 0/1, span 1 got hangup -- Hungup 'Zap/1-1' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: Hottie ?!?
I use the TE110P to connect my Avaya Definity to my * via a TIE/PRI. I just received my two TE411P's. w00T! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Maroney Sent: Wednesday, July 20, 2005 5:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OT: Hottie ?!? Anyone know who that good looking female is thats on the Digium.com website ? Ok, my Real question is I noticed that Digium has relesed a new T1 card with an echo canceller. I also noticed that its supports EM Circuits. Im I have very little knowledge on T1 circuits and traditional PBX's so what Im asking is can I use Digiums T1 card to connect to another PBX via a tie line ? Or does the phone systems have to be the same ? Thank you, Steve Maroney ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Suggestions for using AbsoluteTimeout
I just discovered an 18 hour call to Brazil that was 60 seconds of an employee calling a customer, then 18 hours and 47 minutes of background noise in their office. The Cisco 7960's have an issue where you sometime don't realize the phone is still off hook as was the case for this call. I'd like to use some options like AbsoluteTimeout, which would jump in the middle of a call every 60 minutes, and ask the calling party (for outbound calls only) to press 1 to continue. The problem I find with AbsoluteTimeout is that the call is disconnected BEFORE it reaches the T extension. Is there anyway to temporarily interrupt the call then reconnect the call when the user hits 1? Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bridged-appearances
Has anyone figured out how to mimick a traditional bridged-appearance? My guys like the ability to put a call on hold on line 3 and it's the same call on line 3 on everyone else's phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Strange Inbound Ring Handling
You might look at the r options in the Dial command. Seems like one of these should fit: r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. R: Indicate ringing to the calling party when the called party indicates ringing, pass no audio until answered. This is available only if you are using kapejod's bristuff. m: Provide Music on Hold to the calling party until the called channel answers. This is mutually exclusive with option 'r', obviously. Use m(class) to specify a class for the music on hold. http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Dial -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Dugas Sent: Wednesday, June 15, 2005 9:06 PM To: Asterisk Mailing List Subject: [Asterisk-Users] Strange Inbound Ring Handling Got a wierd one that's reminding me of a problem mentioned in an earlier post but for the life of me, I can't find it. So... Inbound calls via a Voicetronix interface on my Asterisk box are being properly detected and routed to my dialplan as expected. It's a simple plan right now that rings a few internal Voicetronix and SIP stations. When the inbound line rings, it's ringing the internal extensions a couple times then it seems the FXO channel thinks the ringing has stopped so the stations stop ringing too. After a second (or two?), the FXO channel gets the next ring and stars over again. The result is that internal users hear a couple rings, a pause, then another couple rings and a pause again. A Cisco on my desk is showing 24 missed calls when actually there were only a few! Does this ring a bell with anyone (no pun intended)? Paul PS: I'm running CVS HEAD on this test machine. -- Paul Dugas, Computer Engineer Dugas Enterprises, LLC [EMAIL PROTECTED] phone: 404-932-1355 522 Black Canyon Park http://dugas.cc fax: 866-751-6494 Canton, GA 30114 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Remote CDR logging on mysql:
I'm trying to setup remote CDR logging, as directed by: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cdr%20odbc Anyone have example of what I need to change to make an asterisk server log on a remote mysql server? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cmd curl crashes asterisk:
I recently began using the curl cmd to do an external callerid
lookup on my own customer database. I've noticed certain lookups will cause
a crash and not show anything in the messages file or the console. The curl
command is connecting to an external webserver which has a oracle db
connection. The file its hitting is PHP and does a very simply lookup
showing the text like C1234 Bobs mowing service which I later cut off at
15 characters to squeeze it in setcidname(). Here is an example crash.
-- Goto (macro-getcustid,s,3)
-- Executing NoOp(Zap/2-1, Call from using .) in new stack
-- Executing Curl(Zap/2-1,
http://old-inside.theplanet.com/xmlservices/cnum_lookup.html?cid=;) in new
stack
pbx01*CLI
Disconnected from Asterisk server
[macro-getcustid]; ${DEFAULT} is my own number..i.e. no cid was given...
exten = s,1,setvar(CURL=)
exten = s,2,gotoif($[${CALLERIDNUM} = ${DEFAULT}]?9:3)
exten = s,3,noop(Call from ${CALLERIDNAME} using ${CALLERIDNUM}.)
exten =
s,4,curl('http://mywebserver-name/xmlservices/cnum_lookup.html\?cid=${CALLER
IDNUM}')
exten = s,5,setvar(CURL=${CURL:0:15})
exten = s,6,noop(Setting callerid ${CALLERIDNUM} to ${CURL})
exten = s,7,setcidname(${CURL})
exten = s,8,goto(s,10)
exten = s,9,noop(Skipping because CID = ${CALLERIDNUM})
exten = s,10,noop
I can easily avoid these crashes (I hope) by not executing the curl
command if the ${CALLERID} variable is less than 10 characters, but I
thought I would point out that CURL should not be crashing the whole system
because a URL was disliked.
Asterisk CVS-HEAD-04/14/05-15:57:59
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RE: [Asterisk-Users] CallerID
If everyone running windoze had a xserver running, it would be easy... Just
have the * display a window on the users windoze box.
The most useful command I've found so far, it the curl(URL) command. I use
this to do a lookup on inbound callers ${CALLERIDNUM} and see if matches an
existing customer or employee. If it does, I set the caller name to the
returned customer number and company name. Although the damned 15 character
limit is a real pisser... Anyone know a way around that?
You could use curl to invoke the remote popup, but short of using the
existing jabber interfaces or something, I'm not sure how to invoke window
pops on a windoze box.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Wednesday, May 25, 2005 11:41 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] CallerID
We are mostly trying to figure out ways to enable web apps to display
callerid in realtime and also run crm apps passing parameters, etc.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Matt Riddell
|Sent: Miércoles, 25 de Mayo de 2005 11:02 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] CallerID
|
|Anton Krall wrote:
| Do you know any apps that can receive informatio non tcpip ports and
| display it and maybe run an external app upon receiving
|something as an event?
|
|:)
|
|Did you have something in mind?
|
|Do you want me to write you something?
|
|--
|Cheers,
|
|Matt Riddell
|___
|
|http://www.sineapps.com/news.php (Daily Asterisk News - html)
|http://www.sineapps.com/rssfeed.php (Daily Asterisk News -
|rss) ___
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RE: [Asterisk-Users] Compile problem on last CVS
Maybe try a version of redhat that was released in the past 5 years? Seriously, why do you require RH7.3 over Fedora or even RH 9? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thierry Wehr Sent: Sunday, May 15, 2005 5:58 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Compile problem on last CVS Good evening from the CVS of the 2005/05/14 it's impossible to build asterisk* on a redhat 7.3 i get this at compile time chan_sip.c: In function `build_user': chan_sip.c:10007: parse error before `struct' chan_sip.c:10029: `userflags' undeclared (first use in this function) chan_sip.c:10029: (Each undeclared identifier is reported only once chan_sip.c:10029: for each function it appears in.) chan_sip.c:10029: `mask' undeclared (first use in this function) chan_sip.c:10094: warning: type defaults to `int' in declaration of `__s' chan_sip.c:10094: warning: comparison of distinct pointer types lacks a cast chan_sip.c: In function `build_peer': chan_sip.c:10176: parse error before `struct' chan_sip.c:10221: `peerflags' undeclared (first use in this function) chan_sip.c:10221: `mask' undeclared (first use in this function) chan_sip.c:10391: warning: type defaults to `int' in declaration of `__s' chan_sip.c:10391: warning: comparison of distinct pointer types lacks a cast make[1]: *** [chan_sip.o] Erreur 1 make[1]: Quitte le répertoire `/usr/src/asterisk-cvs/asterisk/channels' make: *** [subdirs] Erreur 1 may be someone have a clue to fix it best rehards Thierry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Road Warrior phone config
Or have a small solar panel on the back of the phone. Stick it on the dash of your car, assuming it doesn't burst into flames from heat; it should be fully charged in an hour or two. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Sent: Sunday, May 15, 2005 6:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Road Warrior phone config Andres Paglayan wrote: question about this thread, would a wi-fi voip phone work for this guy? meaning, he takes it to wherever he goes and it gets registered wherever it as wireless access. is that theoretically correct? I like that approach. Those toys will be getting more affordable. One concern I would have is battery life. I think a wisip phone that can be recharged/powered via standard usb cable would be nice. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXO/FXS suggestions:
Im looking for a zaptel type device with one (or more) FXO and one (or more) FXS port. Basically this guy would sit in-line of your phone line (PCI card). Any suggestions? TDM400 would be overkill. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice outage times?
Has anyone been watching and logging when broadvoice becomes unstable? Is it only peak hours, or is it random? If its somewhat consistant, Id like to enforce some time of day routing in my dialplan. Otherwise I may just close the account altogether ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice outage times?
I don't mind so much that calls fail occasionally, but the fact that Broadvoice will let the failed call ring for 15 or 20 seconds, answer the call, play a cute little call can't be completed message, then hangup...really frustrated me. Why can't they send back a busy or congestion signal like every other telco in the world so I can try to reroute the call on another trunk. Right now, I never see failed attempts because something is answering them! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luki Sent: Saturday, May 14, 2005 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Broadvoice outage times? As far as I noticed, it's mostly random and seems to depend more on the origin/destination of the call rather than time of day. But that's not the point -- you shouldn't have to tweak your dialplan because a service only works sometimes. That's just isn't good enough. --Luki On 5/14/05, Tim Connolly [EMAIL PROTECTED] wrote: Has anyone been watching and logging when broadvoice becomes unstable? Is it only peak hours, or is it random? If its somewhat consistant, I'd like to enforce some time of day routing in my dialplan. Otherwise I may just close the account altogether. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Just added snom Mass Deployment
This could be expanded to about any hardware. I can envision using this instead of a callmanager to provide on the fly Cisco 7960 configs. Good work wiki-ing this. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Sunday, May 08, 2005 1:48 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Just added snom Mass Deployment Just added snom Mass Deployment http://www.voip-info.org/tiki-index.php?page=snom+mass+deployment -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 8+ line receptionist only setup
Can your receptionist handle 6 active conversations? Once she transfers the call, it would disappear from those 6 lines. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Manjit Riat Sent: Sunday, May 08, 2005 5:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] 8+ line receptionist only setup Hi, We are looking towards a 8+ CO line setup (20 extensions) in our office but we do not want an IVR(auto-attendant) feature. All incoming will be answered by a receptionist. I have read the multi-line configuration for cisco 7960 thread in this list but that way I believe we could only display 6 incoming lines. What will happen to the rest? Does the expansion module for the cisco 7960 work with asterisk? Any other phones with more lines displayed for the receptionist. Or another solution to tackle this? Thanx ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Asterisk at home with Broadvoice?
Yeah, Broadvoice sucks, everybody cancel your service so I can use it! I have yet to find another provider with as many free calls from a basic rate with no strings attached (Learn from this Vonage!). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Disgruntled Asterisk Luser Sent: Sunday, May 08, 2005 9:06 PM To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RE: Asterisk at home with Broadvoice? Don't worry about these subtle details. Broadvoice has been off the air for almost a solid week, with no real explanation as to what the problem is. On the voxilla.com board, there have been a lot of inferences, but no real solid information as to what the problem actually is. But I really wonder about the survivability of a telephony service business which is unusable for days on end, and with no explanation of what the problem is. On Sun, 8 May 2005, Luki wrote: 2 hour call does not appear to be 'normal residential use' That is insane. Clue: teenagers. They can spend hours on the phone and that certainly is NOT business use. I am about to cancel my service and demand that ... they waive the cancelation fee I do not think there is a cancellation fee; I had someone cancel about a month ago because their DSL connection quality was useless for VOIP (and this certainly was not Broadvoice's fault) and no fee was charged. Beware, if you cancel 3 days into your new billing cycle, they close your account right away, even if you just paid for a complete month 3 days ago. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] spandsp
Does this path already exist??? /var/spool/asterisk/fax/2201001/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ma Zhiyong Sent: Sunday, May 08, 2005 11:27 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] spandsp Hi, I installed spandsp and test it with Eicon card. When fax begin from eicon card to spandsp. It fails and shows: -- Executing RxFAX(Zap/124-1, /var/spool/asterisk/fax/2201001/1115604630.5.tif) in new stack -- Channel 0/31, span 4 got hangup May 9 10:10:41 DEBUG[4967]: app_rxfax.c:246 rxfax_exec: Got hangup Any one has this experiment? How can I get more log infomation about softmodem? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA186 Fax problem solved:
I fought with my ata186 until I decided to start dorking with the settings. I found no outbound faxes could be sent (fax handshake never could complete) until I set the AudioMode 0x00050005. Basically this sets the ATA for fax mode which is documented on: http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administratio n_guide_chapter09186a00801e0dff.html#wp1012620 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] unknown RTP codec 72
I see the same error on codec 100 when I try to rxfax. The faxes fail btw From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomas Sia Sent: Friday, May 06, 2005 12:27 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] unknown RTP codec 72 can anyone tell what is the unknown RTP codec 72 means and how to fix it. I'm using xlite to call PSTN line and the message just pop up on my console but the call can be connected. What am I going to do? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on SMP machine with ztdummy working ?
I've got three dual Xeon's running Redhat Enterprise 4 with 2.6.9 and CVS-HEAD from about a month ago. I didn't have any problems whatsoever, other than the problems I blame on being reluctant to RTFM. No problems with the SMP side whatsoever. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vamsi Pottangi Sent: Friday, May 06, 2005 10:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk on SMP machine with ztdummy working ? Hi All, Was any Asterisk installation on SMP machine successful. Were you able to get ztdummt working on it. If so please let me know which linux favour you are using and any important steps to follow. I have a Dell Power edge 2800 and wanted to try asterisk on it and also use meetme. Which Linux flavour should I go for and the timing source. I don't have a zaptel interface so wanted to use ztdummy. Please guide me. I tried with FC3 as mentioned in below mail but loading of zap module fails saying resource busy. Thanks, ~Vamsi -- Forwarded message -- From: Vamsi Pottangi [EMAIL PROTECTED] Date: May 5, 2005 7:51 PM Subject: chan_zap.so: load_module fails: Fedora Core 3: SMP To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi, I'm trying to install asterisk on Dell power edge 2800 running Fedora core 3. I don't have have any zaptel cards, so trying to use ztdummy. /dev/zap is successfuly created... but I see some problems while starting asterisk ... chan_zap fails to load. Can somebody please help me in overcoming this problem. I was able to run asterisk on other normal PCs running Fedora core 3. Is this something to do with SMP ? I compile zaptel using the link to smp source code only. lrwxrwxrwx 1 root root 34 May 5 21:22 linux-2.6 - /lib/modules/2.6.9-1.667smp/source May 5 21:43:55 VERBOSE[12931]: [chan_zap.so]May 5 21:43:55 VERBOSE[12931]: [chan_zap.so] = (Zapata Telephony) May 5 21:43:55 DEBUG[12931]: Parsing /etc/asterisk/zapata.conf May 5 21:43:55 WARNING[12931]: Unable to specify channel 1: Device or resource busy May 5 21:43:55 ERROR[12931]: Unable to open channel 1: Device or resource busy here = 0, tmp-channel = 1, channel = 1 May 5 21:43:55 ERROR[12931]: Unable to register channel '1' May 5 21:43:55 WARNING[12931]: chan_zap.so: load_module failed, returning -1 May 5 21:43:55 DEBUG[12931]: Unregistering channel type 'Zap' May 5 21:43:55 VERBOSE[12931]: == Unregistered channel type 'Zap' May 5 21:43:55 WARNING[12931]: Loading module chan_zap.so failed! [EMAIL PROTECTED] ~]# uname -a Linux noname11 2.6.9-1.667smp #1 SMP Tue Nov 2 14:59:52 EST 2004 i686 i686 i386 GNU/Linux [EMAIL PROTECTED] ~]# [EMAIL PROTECTED] ~]# ls -l /dev/zap/ total 0 crw--- 1 asterisk asterisk 196, 254 May 5 21:31 channel crw--- 1 asterisk asterisk 196, 0 May 5 21:31 ctl crw--- 1 asterisk asterisk 196, 255 May 5 21:31 pseudo crw--- 1 asterisk asterisk 196, 253 May 5 21:31 timer [EMAIL PROTECTED] ~]# Thanks, ~Vamsi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] broadvoice not hanging up
Me thinks broadvoice needs to add a few more proxies to the US and other hot spots... I'd like to be able to accept calls from any of their proxies. I can see us registering with all 3 and choosing an outbound using lowest latency. yawn okay, I'm out.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luki Sent: Thursday, May 05, 2005 12:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] broadvoice not hanging up Nope, no hangup issues (log says we had over 80 calls total today), however, as of 8pm PST we're back to the Got SIP response 404 Not Found back from 147.135.0.128 issues with registration. I just noticed it... we shall see how long that lasts :-(. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Working exten= fax...
Can someone send me an example of a CVS-head extension.conf excerpt that utilizes the faxdetect and fax extension feature. Im tired of seeing these: Apr 29 17:33:15 NOTICE[3541] chan_zap.c: Fax detected, but no fax extension ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FXO ATA?
Pass through has the same functionality as a modem with a line and a phone connection. Line is where you plug in the dialtone, the dial passes through the phone connection unless the card picks up (like a modem does). I have a X100P clone that is setup as a passthrough. I've never seen a pass through on a FXS, but then I've only messed with ATA-186's recently. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Gabrielson Sent: Thursday, May 05, 2005 9:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FXO ATA? On Thursday 05 May 2005 05:28 pm, Joseph wrote: It has 1-FXS and one 1-Life Line (it is pass through type) I've seen the pass-through term used alot and I'm not quite for sure what that means. What is the difference between a passthrough type and a regular FXO. What can you do with one that you can't do with the other? I noticed that the wiki says that the handytone 486's lifeline FXO port is not usable via SIP, only used as a fallback for power failure. Is this considered a passthrough or are there 3 types, pass-through, lifeline, and full FXO. Thanks, Jon. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for Log parse for CDR's
I know somebody out there has a little perl script that parses the cdr file into calls per hour and calls per month. Anyone want to save me an hour? Please? My wife will thank you! Thanks in advance, Tim image001.gif___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MOH
First off, yes, canreinvite=no would be a good choice. Secondly, did you make mpg123 from the asterisk source directory? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Wednesday, May 04, 2005 10:49 PM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] MOH I just set up * on a new server, MOH does not work. I just figured out that when I allow two endpoints to connect directly (canreinvite=yes) MOH does not work. It seems the reinvite on hold back to Asterisk either is not supported by Asterisk or is not working in this case - any idea which one it is? Nabeel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P Drops Calls after many touch tones fromcaller
Do you have dial command in there with option t or T? Whats the log say right before a call is dropped ? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barton Fisher Sent: Wednesday, May 04, 2005 11:13 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] TE410P Drops Calls after many touch tones fromcaller Ihave a TE410P card with two Telco T1's and two external IVR systems attached. Calls from Telco are routed to proper IVR system based on DNIS (DID) received from Telco using a native bridge. T1's are D4 AMI SF Some IVR applications requires the caller to enter digits using their touch tone phone such as phone number. Not every time, but enough to be annoying asterisk drops the call after about 6 - 10 digits. I've adjusted to busy count to 15 and turn off busy detect, but it still happens. I removed asterisk and attach directly to IVR, it works. What I'd like to do is have asterisk only monitor the port for hang-up, not listen for touch tone if that's what causing the problem Any ideas to try would be appreciated Bart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIPP with asterisk
How about: exten = 9111222,1,answer exten = 9111222,2,wait(10) exten = 9111222,3,Hangup -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tulika Pradhan Sent: Wednesday, May 04, 2005 11:29 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIPP with asterisk i am trying to do load testing on asterisk using sipp testing tool. i am able to send invite requests to asterisk by using sipp -sn uac ip address -s 9111222 -d 1 -r 10 i am also running sipp -sn uas on the same box but no message arrives on uas part. and asterisk returns error while dialing. Unable to create channel of type SIP the extensions.conf has exten = 9111222,1,Dial(SIP/9111222) exten = 9111222,2,Hangup what config changes should i make to answer the calls landing on asterisk ? tulika _ Insta predictions! http://www.astroyogi.com/newmsn/astroshopping/astrologerservices/express.asp Get your answers in 48 hours! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem with realtime SIP
Let's see your sip.conf and a sip show users. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Callum McGillivray Sent: Wednesday, May 04, 2005 11:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Problem with realtime SIP Hi Guys, We have just set up Asterisk (CVS Head) for a realtime enviorment using MySQL Asterisk Addons. I have populated the sip_buddies table with the same information that is came from our sip.conf, however registration seems to fail for the softphone we have set up. Does anyone have any idea as to what I should be looking for here? I'm not getting any error messages in debug, and just this line from the command line; May 5 14:30:18 NOTICE[5063]: chan_sip.c:9020 handle_request_register: Registration from 'Callum McGillivraysip:[EMAIL PROTECTED]' failed for '192.168.1.90' Can someone tell me what I might be missing ? Or can someone give me a dump of their sip_buddies table so I can try and see what I might be doing wrong ? Thanks, Callum ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MOH
Somebody correct me if I'm wrong here, but without reinvite being disabled, I don't think the * can inject audio on the middle of the call. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Wednesday, May 04, 2005 11:41 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] MOH First off, yes, canreinvite=no would be a good choice. Well, I am in a situation where my * server is hosted and it is quite pointless to have all media going through the * server when two SIP devices are talking. Secondly, did you make mpg123 from the asterisk source directory? Yes, tried that. It said mpg123 is up to date. -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 FWD: 46990 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SNMP Monitoring
I use MRTG to graph Active/Configured SIP channels and Active/Total PRI/ZAP channels, but I don't monitor the up/down status. You could probably write a little perl script to tail the logfile and watch for certain events, then forward them by mail. Actually, I think I might do that too since I've only got one active PRI at the moment. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Tuesday, May 03, 2005 10:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SNMP Monitoring I've read on the wiki how you can SNMP monitor an Asterisk machine and from what I read, you're pretty much monitoring the availability of Asterisk. I'm looking for a way to be able to monitor the availability of individual T1 circuits of my TE410P card. During the storm season, some of our T1s tend to flap and I'd like to be able to monitor that. Is there something that can do this? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to cancel a transfer in progress:
Is there a feature code you can dial after beginning an atxfer (*2) that will bail out and return you to the caller. Let's say I want to transfer to the CEO of the company, but only if he is available. Once I hit *2, punch in his extension, I don't of anyway to cancel out. If I hit * or hangup, the transfer completes anyway. No other keys seem to do anything once the transfer has started. I saw one person asking the same thing in a comment on the wiki page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20features.co nf#comments Anyone have an answer, or does this need to be added? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bug found in SJLabs SJPhone concerning dialpad
Seems as though the dialpad in SJPhone cannot me used to signal *. *2 doesn't do anything except play a DTMF in your ear. If you use your keyboard to send shift-8, 2, all works as expected. Bug report submitted already. Cheers Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Choppy Sound on PSTN End
Title: Choppy Sound on PSTN End I have the exact setup you describe, SJPhone - * - Zap/PRI. I think you need to twiddle some settings. You might turn on qualify just to see if the * is seeing network flaws. Keep in mind, if your using windows, anytime the user starts clicking around, you can expect less than ideal audio. Also, why disable GSM ? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim ChandlerSent: Monday, May 02, 2005 11:23 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Choppy Sound on PSTN End Hi all, I recently set up Asterisk on a Dell PowerEdge 1850 Dual Xeon 2.8 Ghz processor. I am running the latest build of White Box Enterprise Linux. Our call routing is like this: SJPHONE on Windows - QoS-enabled Switch - Asterisk - T1 Line - Broadvoice SIP account - PSTN Calls seem to work great from user to user. However, calls from a SJPhone user to the PSTN are not so great. The SJPhone user hears the person on the PSTN perfectly I mean, completely flawless. However, the user on the PSTN end hears choppy / jittery, extraneous clicks, etc. Here is the SJPhone config: Audio Compression: G.711 Driver buffer size: 20 msec Driver input queue length: 6 Driver output queue length: 4 RTP jitter queue length: 6 Silence Suppression: No DTMF Sending: RFC 2833 Signal Duration (ms): 270 RTP Payload type: 101 Signal volume: 10 Pause duration (ms): 100 And the sip extension config (in Asterisk Management Portal): Allow: blank Canreinvite: no Disallow: gsm Dtmfmode: rfc2833 Host: dynamic Nat: yes (some users are behind NAT) Qualify: no Any ideas on what to do to get rid of the choppiness? Thanks! Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Things to backup:
I'm about to add a new wiki page, but wanted some input. This is a list of locations where asterisk specific files are located. In my case, this is RHES4 specific, I'm telling my backup software to backup: /etc /usr/src (yes I know, but there is a lot of custom crap in there) /usr/lib/asterisk /usr/include/asterisk /var/lib/asterisk /var/spool/asterisk /var/log /usr/sbin/ast* /var/www Anything else I'm missing??? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail volume with sipura 3000
Use wav, not gsm or wav49. /etc/asterisk/voicemail.conf ; ; Voicemail Configuration ; [general] format=wav -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of snacktime Sent: Monday, May 02, 2005 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] voicemail volume with sipura 3000 I have my sipura SPA-3000 setup so that incoming calls on the FXO ring for 20 seconds then using dialplan 2 it dials my * server and drops the person into voicemail. The problem is that voicemail messages are way too quiet. You can't hear enough to understand what is being said. If someone answers the phone before the 20 seconds the volume is fine. If I dial from the sipura line 1 into voicemail and leave a message the volume it just fine also. I know most likely this isn't an * issue, but I thought maybe someone else has encountered this problem? Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel
http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy This will help. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Huddleston, Robert Sent: Monday, May 02, 2005 1:26 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Zaptel Forgive my ignorance - I'm building * on an old redhat 8 box... I can't build Zaptel - I don't need libpri - Asterisk is building though. Do I need Zaptel - I remember once using ztdummy or something like that... If I need Zaptel - any ideas why no build.. I'm assuming the common answer is going to be I need to upgrade the Kernel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail volume with sipura 3000
Dunno.. Guess somewhere in the translation it gets amplified to an acceptable level. It seems to work though. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of snacktime Sent: Monday, May 02, 2005 2:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voicemail volume with sipura 3000 On 5/2/05, Tim Connolly [EMAIL PROTECTED] wrote: Use wav, not gsm or wav49. /etc/asterisk/voicemail.conf ; ; Voicemail Configuration ; [general] format=wav For future reference, why is this? Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
