Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *
Hi, Except that for some users 1.2.18 is NOT stable. I've had to roll back to 1.2.15 on my production servers in order to prevent core dumps at least once per day. No, I am not willing to turn my production servers into testing servers to solve this. Doing so would make me a "former consultant" for these customers. I've got some core dump on 1.2.18 and the patch available on ticket 9602 have fix all issues, using 1.2.18 on lots of server without any issues http://bugs.digium.com/view.php?id=9602 -- Joel Vandal ScopServ Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with zaptel drivers or card
Hi, How did you, or do go about reversing the patch? I have put the patch (simple) available at : http://www.scopserv.com/download/patches/zaptel-1.2.12-reverse7860.patch Go on your zaptel src directory and do : patch -p0 < zaptel-1.2.12-reverse7860.patch It is a T1 and I am not sure what you mean by behaves like an E1. The connection is a T1 with 23 b-channels and 1 d-channel. I think it just so happens that the problem channel is 16 on the card. This worked fine for over a year before the upgrade to the zaptel drivers. I`ve got similar problem and look like the patch #7860 is responsable of this issue... like if this patch doesnt check if the line is an E1 or T1. I have reverse the patch on 1.2.12 and all work perfectly now. -- Joel Vandal, CTO ScopServ Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with zaptel drivers or card
Administrator a écrit : It is a T1 and I am not sure what you mean by behaves like an E1. The connection is a T1 with 23 b-channels and 1 d-channel. I think it just so happens that the problem channel is 16 on the card. This worked fine for over a year before the upgrade to the zaptel drivers. I`ve got similar problem and look like the patch #7860 is responsable of this issue... like if this patch doesnt check if the line is an E1 or T1. I have reverse the patch on 1.2.12 and all work perfectly now. -- Joel Vandal, CTO ScopServ Inc. ___ --Bandwidth 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] Find-Me/Follow-ME
Hi Roger , Has anyone developed a web interface where users could setup their own find-me/follow-me services? Yes, this is available on the ScopServ Telephony GUI (Commercial). -- Joel Vandal, CTO ScopServ Inc. http://www.scopserv.com. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7970 Configs
Hi,
I just download the SIP image (cmterm-7970_7971-sip.8-0-2-0.cop) from
Cisco, copy all files on my tftpboot, create a SEP{mac}.cnf.xml file
(take the one posted by Greg Oliver) with some modification.
If the secret= is empty on the server, I receive now request on the
Asterisk server but the phone send the request 2-3 times per second to
the server.
(Repeated request...)
-- Registered SIP '1009' at xx.xx.xx.247 port 49504 expires 3600
-- Registered SIP '1009' at xx.xx.xx.247 port 49505 expires 3600
-- Registered SIP '1009' at xx.xx.xx.247 port 49506 expires 3600
(.)
(Register Request)
REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.0.136:5060;branch=z9hG4bK4dc6894b
From: ;tag=0015f97f42710003b4619858-cc0ebb79
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
Max-Forwards: 70
Date: Mon, 27 Feb 2006 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP7970G/8.0
Contact:
;+sip.instance="";+u.sip!model.ccm.cisco.com="30006"
Since my phone is behind a NAT, I have enable these setting:
true
true
I can establish an SSH connection to the phone (sshUserID/sshPassword)
and can log into phone (debug/debug and log/log) to get more
informations (like show config)
For SIP Proxy Authentication, with "show config", I see a setting for
authPassword, but try to put it on the xml file but doesnt work.
I have never used CallManager, I presume that we need this to generate a
template "SEP cnf.xml" files ? Not found any documentations on Cisco
site about SIP or SCCP parameters that must be in this file.
--
Joel Vandal, CTO
ScopServ Inc.
http://www.scopserv.com/
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Re: [Asterisk-Users] mpg123 alternative?
Hi, I still seem to have the usual two mpg123 processes running with 1.2.4, with whatever music on hold is set in musiconhold.conf I'm sure it is very obvious, but I can't for the life of me figure out what I'm supposed to do to use the built-in MP3 player facilities. [default] mode=mp3 directory=/var/lib/asterisk/mohmp3 random=yes Change mode=mp3 to mode=files then do a "moh reload" on CLI -- Joel Vandal ScopServ Inc. http://www.scopserv.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma analog cards?
Hi Michael, Does anyone on-list have direct experience with the new analog cards from Sangoma? I'm thinking about FXOs with echo cans. Need 2-4 ports but don't want to go through another TDM400 style experience. Next week, I will soon receive some Sangoma Analog and Digital cards to test in order to see if it can fix some echo problems. We will do major testing on theses card because we work on an appliance and hope that this will solve some echo problem that we get with non-sangoma cards. Will be able to give you more information in 1-2 weeks. -- Joel Vandal ScopServ Inc. www.scopserv.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue - check agent
Hi,
I have defined 4 queue's. Is there any way to check is there any agent logged
in any of those queue's?
What I would like to do is to check if there is any agent in any of queue's and
if there is, then I'll will transfer a call to that queue, it there isn't I
would like to do something else with a call.
The Queue application sets the QUEUESTATUS channel variable upon
completion. The status of the call can be : TIMEOUT, FULL, JOINEMPTY,
LEAVEEMPTY, JOINUNAVAIL or LEAVEUNAVAIL.
Here an example
...
exten => 3,5,Queue(scopserv-test|tH|||30)
exten => 3,6,GotoIf($["${QUEUESTATUS}" = "JOINEMPTY"]?1000)
exten => 3,7,GotoIf($["${QUEUESTATUS}" = "JOINUNAVAIL"]?1000)
exten => 3,8,GotoIf($["${QUEUESTATUS}" = "FULL"]?1000)
exten => 3,9,NoOp(Normal Queue exist)
exten => 3,10,Hangup
exten=> 3,1000,Voicemail([EMAIL PROTECTED])
--
Joel Vandal, CTO
ScopServ Inc.
http://www.scopserv.com/
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Re: [Asterisk-Users] loading zaptel drivers automatically upon reboot
Carlos Alperin a écrit : That is right for zaptel. But you still has to do modprobe wctdm on rc.local before to load asterisk. Any way to fix this? On Redhat / Centos / Fedora I usualy do : cd zaptel ; make config cd ../asterisk ; make config chkconfig zaptel on chkconfig asterisk on At boot, it will start zaptel then asterisk. -- Joel Vandal, CTO ScopServ Inc. http://www.scopserv.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] loading zaptel drivers automatically upon reboot
Hi, Just installed Asterisk 1.2 on a brand new clean machine running RedHat 9.0. I have a TDM400 card inside. When I boot, the card seems dead. When I do: modprobe wctdm modprobe Zaptel Since you use Redhat, From zaptel src directory, do a "make config", it will create init.d and sysconfig file for zaptel. You will be able to control zaptel using chkconfig and service (service zaptel stop/start). Same thing for Asterisk. (make config). -- Joel Vandal, CTO ScopServ Inc. http://www.scopserv.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Per Extension Password for Outgoing Routing
Hi, For default, the extensions only can dial to local numbers, but when they want to call to cell phones, long or international phones, there are authorized users, each one with their own password for dialing. I've checked the password for outgoing routing in Asterisk, but the password it's the same for everyone, or i am wrong? And the second issue, the extensions for default only can dial from 7.30am to 4.45pm (office hours); after that, nobody can dial out; but there are users which with a special sequence can dial out. is there a way to implement that functionality in Asterisk? We have implement a similar function in the ScopServ GUI, this feature is HotDesk. For outgoing call, it use an AGI (exten => _X.,1,AGI,scopserv_hotdesk),this is a script that look in a db for Time Schedule, dial permissions. and will ask for a password if required. This script also allow "roaming extension". AGI : 1- Verify if channel have permission (look in an SQL db) 2- Ask for a Password if requiored 3- DBGet channel/TimeSegment (ex: 08:00-17:00|mon-fri) 4- Parse TimeSegment 5- Execute Dial or Fail -- Joel Vandal ScopServ Inc. http://www.scopserv.com \\ Web GUI for Asterisk PBX // ___ --Bandwidth 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] Complete Removal of Asterisk
Hi, Is there any documentation for the complete removal of Asterisk from a Linux/Unix system? I want to install a fresh copy of asterisk. Depend of your distro, You can use emerge for Gentoo or rpm with CentOS/RHEL/Fedora. Debian also have dpkg command. If you have installed from source, you can do a : rm -rf /usr/lib/asterisk rm -rf /var/lib/asterisk rm -rf /usr/sbin/asterisk If you want to remove configs files: rm -rf /etc/asterisk If you want to delete Voicemail and Outgoing Call queues: rm -rf /var/spool/asterisk Thanks, -- Joel Vandal ScopServ Inc. http://www.scopserv.com/ Complete Web GUI for Asterisk PBX ___ --Bandwidth 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 1.0.9 + spandsp 0.0.2pre20 = crash on boot
Hi, I have asterisk 1.0.9 installed with spandsp 0.0.2pre20. Asterisk crashes on boot while loading app_txfax.so & app_rxfax.so. If I move the files out of /usr/lib/asterisk/modules asterisk boots fine. Running on FC3, Linux asterisk.crocker.com 2.6.11-1.27_FC3smp #1 SMP Tue May 17 20:43:11 EDT 2005 i686 i686 i386 GNU/Linux Edit " /etc/ld.so.conf " file, add the " /usr/local/lib " directory then do " ldconfig " Asterisk doesnt start because app_tx/rx miss a library. -- Joel Vandal ScopServ Inc. http://www.scopserv.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pager Notification Script
Tom Rymes a écrit : Does anyone on the list have a script for notifying pagers that they would be willing to share? I have found a reference in the archive to such a script, but previous attempts to find the author of that posting have failed. Anyhow, I am looking to set up a system whereby a message is sent to a pager when a voicemail is left in a specified mailbox. (This is easy, it's built-in to Asterisk). Then, if that message hasn't been retrieved in 5 minutes, I want to send another page. The same goes after 10 and 15 minutes. After 20 minutes, I want to send another page *AND* send an e-mail or generate a call to another party. Off Site Notification or Off Premise Notification... I have write a script that is part of ScopServ but here how it work: - Create per-user configs using GUI (ex. after 10 min send to a voicemail, after 20 min. send to a pager, etc) (email, pager, voicemail) - Use externnotify in voicemail.conf - If # of msg = 0 then delete all pending notification else - Retreive per-user config and check action - Create action in a second table with timestamp + x min. - A crontab that check at each minute for action, execute if and delete the row in table. - Create .call file or send email -- Joel Vandal ScopServ Inc. http://www.scopserv Inc. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Complete NPA-NXX list for USA/Canada npanxx, ratecenters, etc (attached)
Hi, - Original Message - Since we are all trading secrets, check this site out http://members.dandy.net/~czg/lca_index.php You can get this Perl scripts that extract NPA-NXX directly from dandy.net... http://www.voip-info.org/tiki-index.php?page=ScopServ%20LCA%20Map -- Joel Vandal ScopServ Inc. ___ --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] TE406P & TE411P - Channelized voice and data T1/E1 PCI cards with Echo Cancellation
Hi, Someone have more info about TE406P/TE411P ? (from VON europe mailing) Stop by the Digium | Asterisk booth #1151 and check out the latest new Digium products, including: a.. Asterisk Business Edition b.. DS3000P - Channelized DS3/T3/E3 voice and data PCI card c.. TE406P & TE411P - Channelized voice and data T1/E1 PCI cards with Echo Cancellation -- Joel Vandal ___ 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] Flashpannel: How to get more than 28 buttons?
Hi, I also wrote a PHP scripts that generate op_style.cfg. You specify how many rows x cols and the icons/buttons/text alignment are properly scaled. (i.e. you defined a 5 x 20 for 100 buttons, button height will be small so "Line", "CallerID", "Timer" position will be "adjusted") Script not 100% finish but will be available soon... -- Joel Vandal - Original Message - From: "Robert Rozman" <[EMAIL PROTECTED]> To: "Nicolás Gudiño" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, March 16, 2005 8:10 AM Subject: Re: [Asterisk-Users] Flashpannel: How to get more than 28 buttons? Hi, I'd also like to see alternative op_style.cfg. Can we establish some place to share them ? I've also one with smaller buttons (but will have to count them :-) ... Regards, Rob. - Original Message - From: "Nicolás Gudiño" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, March 16, 2005 1:26 PM Subject: Re: [Asterisk-Users] Flashpannel: How to get more than 28 buttons? Hi Ronald, I have setup flash pannel, ... looks nice, but so far I could not configure it to get more than 4x7 buttons. I tried to make the buttons smaller, but than just the entire picture is smaller. What did you change in op_style.cfg? You can have literally hundred of buttons per screen, or multiple 'context' to split your buttons into several screens. I wll send you an alternate op_style.cfg with smaller buttons offlist. Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@Home 0.6 Released [Follow Me]
Hi, Does it's possible to get more information about your design ? Thanks, -- Joel Vandal - Original Message - From: "Race Vanderdecken" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]>; "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Thursday, February 17, 2005 5:10 PM Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] 0.6 Released [Follow Me] I have a design that works for "Follow-Me" and "Find-Me" is anyone is interested. I can help you with the code, but don't ask me to check-it in to the CVS. Race "The Tyrant" Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nitesh Divecha Sent: Thursday, February 17, 2005 2:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] 0.6 Released [Follow Me] Hello All, And thanks for the [EMAIL PROTECTED] 0.6! It works awesome. Any plans to implement "Follow Me" feature? Nitesh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MultiLine Sip Phones (3com 3102)
Hi, I've just get a 3COM 3102 but is not configured to use SIP protocol. I've read that I need an NCP PBX from 3com to upgrade to SIP firmware ? Does it's true ? I must try to upgrade this =) If someone can help me... Thanks. -- Joel - Original Message - From: "James Bean" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, February 19, 2005 2:17 AM Subject: RE: [Asterisk-Users] MultiLine Sip Phones No unfortunately a lot of the extensions do not have PC's near them or in there offices, and the people involved are a little on the computer illiterate side, although I am slowly training them. They just want a phone that shows them extensions/lines and who is using them That's why I am hoping someone else has used the 3Com Business Phone 3102 as it comes standard with 18 function keys, just hoping they work the same way as the snom. James ___ 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] Meetme
Can someone see what's wrong here please ? exten => 550,1,Answer exten => 550,2,Wait(1) exten => 550,4,MeetMe(18|Md) exten => 550,5,Hangup The priority 3 is missing ... 1, 2 then timeout... -- Joel Vandal ScopServ Inc. ___ 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] 191st simultaneous call fails
What about the limit of 200 Zap channels ? The server doesnt want to create the channel 201... - Original Message - From: "Nick Bachmann" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Saturday, December 18, 2004 6:23 PM Subject: Re: [Asterisk-Users] 191st simultaneous call fails Jim Gottlieb wrote: I've been testing both T400P and TE405P boards and I'm running into some kind of hard limit on the number of simultaneous calls. This is on x86 with 2 Athlon MP 2800+ CPUs running Fedora Core 1. Everything is fine up to 190 channels, but the 191st call fails every time with errors like: Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on Zap/201-1 Dec 14 15:44:00 WARNING[1215]: Failed to create update thread! Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on channel 0/9, span 9 Dec 14 15:44:00 WARNING[1215]: Call specified, but not found? Dec 14 15:44:00 WARNING[1215]: Hangup on bad channel 0/9 on span 9 It's not tied to which channel the call comes in on. It's some resource that's exhausted after 190 calls. A limit on threads? Try http://people.redhat.com/alikins/tuning_utils/thread-limit.c and see what happens. Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] Softphone for PocketPC or iPaq
Try SJPhone... - Original Message - From: "Sudhir Kumar" <[EMAIL PROTECTED]> Subject: [Asterisk-Dev] Softphone for PocketPC or iPaq ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Convert Cisco 7960 to sip
All Cisco 7940 that I have upgrade to 7.1 no more try to get the dialplan and ringlist files from tftp. Now I must found a way to "downgrade" from 7.1 to 6.3. -- Joel - Original Message - From: "Simon Brown" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] Convert Cisco 7960 to sip I've been using V7 for a couple of months now with no problems. Simon Brown ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Lots of FXS ports / Channel Bank ?
Hi, I have a client that have currently 400 analog phones (all wired w/ Cat3). I need multi-ports FXS interfaces but I only find 24 ports FXS (like Mediatrix 1124) but it's a little bit expensive to get 15-16 box (~408 FXS ports). Someone have suggestion to link all theses phones to "channel bank" ? What equipment to use ? Currently all 400 lines are terminated in M-66 blocks and want to connect all these w/ an RJ21 multiports FXS ? -- Joel Vandal
