Re: [asterisk-users] How to detect fake CallerID? (8xx?)

On Wed, May 10, 2017 at 10:11 AM, Steve Edwards wrote: > I have a 'time and attendance' application. Think janitorial or security > kind of thing where an employee goes from location to location. > > They're supposed to 'clock in' when they get to a site using a phone

Re: [asterisk-users] Script to Program Snom Phones

On Thu, Apr 9, 2015 at 12:23 PM, Derek Andrew derek.and...@usask.ca wrote: SNOM phones can be configured using files on a TFTP server. On Thu, Apr 9, 2015 at 11:14 AM, jg webaccounts...@jgoettgens.de wrote: Does anyone know how to program Snom phones using a Mac addresses like what is

Re: [asterisk-users] recommendations for RJ-11 surge supressors?

On Thu, Jun 27, 2013 at 10:34 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Thursday 27 June 2013, Eric Cooper wrote: I'd like to protect my expensive Digium FXO cards from spikes on my three incoming PSTN lines. Does anyone have any recommendations? Does your telco not fit surge

Re: [asterisk-users] Optimizing Asterisk Environment

On Sat, Mar 23, 2013 at 12:06 PM, Joshua Colp jc...@digium.com wrote: Nick Khamis wrote: Oh no secret. Some things I do is increase the ulimit size. I was wondering if there was a way to increase allocated memory. I have been reading about a -p option but when I start asterisk using asterisk

Re: [asterisk-users] Optimizing Asterisk Environment

protocol. Please let us know of some digium solutions. Again, we would love to support the cause. Nick. On 3/23/13, Andrew Latham lath...@gmail.com wrote: On Sat, Mar 23, 2013 at 12:06 PM, Joshua Colp jc...@digium.com wrote: Nick Khamis wrote: Oh no secret. Some things I do is increase

Re: [asterisk-users] Asterisk authentication on LDAP (SSHA and SHA passwords)

On Sun, Mar 10, 2013 at 11:37 AM, Paulo Victor Fernandes da Silva paulovictorsi...@gmail.com wrote: hello guys, I'm working on a federal university at Brasil, we already have an openLdap with all users and this base is used to authenticate several services like email, vpn, wireless

Re: [asterisk-users] Conf Bridge

On Thu, Jan 17, 2013 at 3:02 PM, Bryant Zimmerman brya...@zktech.com wrote: Hey all. RE: Conf Bridge. I am looking into a project that would need 8 to 10 thousand parties in a single conference. Most would be on mute but 5 to 6 would be presenters. Is the new conf bridge solid enough to

Re: [asterisk-users] Need Help

On Thu, Jan 17, 2013 at 3:05 PM, Joe Ruffolo j...@mrkgroup.com wrote: Hi all! In need of some serious help. We currently run Trixbox and Cent Os on a 2u server for our small business’s phones system. We are using some Polycom Soundpoint IP phones. The whole thing came crashing down over

Re: [asterisk-users] Mail list settings?

On Thu, Jan 17, 2013 at 6:32 PM, Bryant Zimmerman brya...@zktech.com wrote: Hey all For some reason the mailing list is sending all messages from the sending party. This makes it less than ideal when responding; as selecting reply goes to the person and not the list. Can we have it set back

Re: [asterisk-users] Asterisk Support from Digium

On Sat, Nov 3, 2012 at 2:16 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello, I wonder if Digium provides support for Asterisk OpenSource versions as an anual fee or something? For example, if i download Asterisk 1.8.X (Certified or not...) can i buy support from Digium to maintain and

Re: [asterisk-users] Asterisk error message so uncommon, not even Google knows abuot it

On Fri, Oct 19, 2012 at 11:28 AM, Eric Wieling ewiel...@nyigc.com wrote: I'm setting up a test server with a Digium TE122 and am getting the following error on the console, spewing as fast as it can. Does anyone have any idea what this error might be? [Oct 19 11:24:53] NOTICE[2076]:

Re: [asterisk-users] asterisk installation under a single directory

On Mon, Oct 15, 2012 at 3:07 AM, sudeep melekar sudeep.meleka...@gmail.com wrote: hello, i want to install asterisk 1.8 in a single directory myasterisksetup i.e after asterisk installs it put some of it's installation files in different directories e.g /var/log/asterisk /var/run/asterisk

Re: [asterisk-users] Parameterize asterisk config files

On Tue, Oct 2, 2012 at 8:04 PM, Steve Edwards asterisk@sedwards.com wrote: On Tue, 2 Oct 2012, Mitch Claborn wrote: I'd like to be able to use the same config files in CVS and have the differences resolved at run time, based on host name of the asterisk server. Another idea would be to

Re: [asterisk-users] 'Training mode'

On Fri, Sep 28, 2012 at 5:27 PM, Adam Moffett adamli...@plexicomm.net wrote: I was asked today if we could somehow have a trainee on the phone with a supervisor conferenced in, but somehow have it so anything the supervisor says is only heard by the trainee and not the customer. Is there a

Re: [asterisk-users] How do you convert your prompts to an asterisk-friendly format?

On Tue, Aug 28, 2012 at 10:39 AM, Johan Wilfer li...@jttech.se wrote: Hi, I've used the shells-script at the end of this email to generate 8khz mono wave-files for asterisk from a 144 khz recording. The script does two things: resample normalize the audio volume. Anyone like to share

Re: [asterisk-users] How do you convert your prompts to an asterisk-friendly format?

On Tue, Aug 28, 2012 at 11:00 AM, Johan Wilfer li...@jttech.se wrote: 2012-08-28 16:44, Andrew Latham skrev: On Tue, Aug 28, 2012 at 10:39 AM, Johan Wilfer li...@jttech.se wrote: Hi, I've used the shells-script at the end of this email to generate 8khz mono wave-files for asterisk from

Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?

On Mon, Aug 20, 2012 at 10:52 AM, Joshua Colp jc...@digium.com wrote: - Original Message - Joshua Can you copy and past into a wiki page for everyone's benefit? Maybe https://wiki.asterisk.org/wiki/display/~jcolp/WebRTC_Demo_Setup or like page would be good. If this thread has

Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?

On Mon, Aug 20, 2012 at 2:58 PM, Juan Castro jcas...@instant.com.br wrote: Hoo-hah. It registers. Progress! Now... media. Or not. On Mon, Aug 20, 2012 at 12:51 PM, Joshua Colp jc...@digium.com wrote: - Original Message - The complete URL to use is http://asterisk IP address or

Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?

On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro jcas...@instant.com.br wrote: I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Greetings, -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado

Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?

On Fri, Aug 17, 2012 at 4:01 PM, Joshua Colp jc...@digium.com wrote: - Original Message - On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro jcas...@instant.com.br wrote: I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80.

Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?

On Fri, Aug 17, 2012 at 4:07 PM, Andrew Latham lath...@gmail.com wrote: On Fri, Aug 17, 2012 at 4:01 PM, Joshua Colp jc...@digium.com wrote: - Original Message - On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro jcas...@instant.com.br wrote: I see no indication of how to do

Re: [asterisk-users] Websockets on Asterisk 11 and SipML5

james.mortensen at a-cti.com Andrew Latham lathama at gmail.com writes: On Mon, Aug 13, 2012 at 2:58 PM, James Mortensen james.mortensen at a-cti.com wrote: Hello, I'm trying to register a user using sipml5 on Asterisk 11. I followed the instructions here: http://thr3ads.net

Re: [asterisk-users] Websockets on Asterisk 11 and SipML5

On Mon, Aug 13, 2012 at 2:58 PM, James Mortensen james.morten...@a-cti.com wrote: Hello, I'm trying to register a user using sipml5 on Asterisk 11. I followed the instructions here: http://thr3ads.net/asterisk-users/2012/08/1972342-Asterisk-Websockets I added transport=ws to my sip.conf

Re: [asterisk-users] TDM400P: Lifetime Replacement

On Sun, May 6, 2012 at 12:42 PM, Greg Woods g...@gregandeva.net wrote: I have a Digium TDM400P card that appears to have died. The first noted symptoms were that dahdi would fail to reload on boot. On closer inspection, the card looks totally dead; no lights on at all. I have tried moving it

Re: [asterisk-users] Asterisk generating backtrace

On Wed, Mar 21, 2012 at 3:04 PM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, when generating backtrace I get following output : [root@sip ~]# gdb -se asterisk -ex bt full -ex thread apply all bt --batch -c core.sip-2012-03-21T10\:57\:29+0100 /root/backtrace.txt asterisk: No such

Re: [asterisk-users] fallback to default extension

On Wed, Mar 21, 2012 at 8:27 AM, Paolo Supino paolo.sup...@gmail.com wrote: Hi  I was asked by our development departement to setup asterisk in a manner that if someone calls an extension in the department that was was only configured, but a handset was never attached to it to fall back to a

Re: [asterisk-users] 8-span TE820 card and interrupts

On Wed, Mar 21, 2012 at 8:45 AM, Tony Mountifield t...@softins.co.uk wrote: Over the years I have experienced a few interrupt issues when using some of the Digium E1/T1 cards with Zaptel drivers, and usually resolved them by disabling USB devices in the motherboard BIOS settings. Now more and

Re: [asterisk-users] fallback to default extension

On Wed, Mar 21, 2012 at 3:10 PM, Paolo Supino paolo.sup...@gmail.com wrote: H Andrew Your solution is the simplest I received and so I tried implementing it only to discover that it doesn't work as expected... TIA Paolo snip Check your Dial() options... Verify your options to you

Re: [asterisk-users] res_http_post.so questions

On Mon, Feb 6, 2012 at 8:24 PM, Josh mojo1...@privatedemail.net wrote: In short - is this module essential for the running of Asterisk? What is its function? Is there a help/list where I could find a description of what it does? Thanks! The primary goal was to upload audio for IVRs in the

Re: [asterisk-users] Asterisk 1.8 warns for lines starting with # in /etc/dahdi/system.conf

On Thu, Dec 22, 2011 at 2:33 PM, Olivier oza_4...@yahoo.fr wrote: Hi, Testing 1.8.8.0 (with Dahdi 2.5.0.2 and asterisk-gui 2.1.0-rc1), I'm seeing this on my console: WARNING[25363]: config.c:1208 process_text_line: Unknown directive '#' at line 1 of /etc/asterisk/../dahdi/system.conf This

Re: [asterisk-users] Asterisk 1.8 warns for lines starting with # in /etc/dahdi/system.conf

On Thu, Dec 22, 2011 at 2:48 PM, Shaun Ruffell sruff...@digium.com wrote: On Thu, Dec 22, 2011 at 06:33:44PM +0100, Olivier wrote: Hi, Testing 1.8.8.0 (with Dahdi 2.5.0.2 and asterisk-gui 2.1.0-rc1), I'm seeing this on my console: WARNING[25363]: config.c:1208 process_text_line: Unknown

Re: [asterisk-users] Asterisk 1.8 warns for lines starting with # in /etc/dahdi/system.conf

On Thu, Dec 22, 2011 at 3:42 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 12/22/2011 12:02 PM, Shaun Ruffell wrote: On Thu, Dec 22, 2011 at 02:54:05PM -0300, Andrew Latham wrote: On Thu, Dec 22, 2011 at 2:48 PM, Shaun Ruffellsruff...@digium.com  wrote: I could be mistaken, but I

Re: [asterisk-users] Suppress -- Remote UNIX connection message

On Wed, Dec 21, 2011 at 12:03 PM, Bryant Zimmerman brya...@zktech.com wrote: We have written some monitoring and stat collection scripts that use asterisk -rx command  The script runs once a min and logs data and posts any critical notifications.  Everything is working well with this method but

Re: [asterisk-users] Digium TE205P leds flash red on startup

On Thu, Dec 15, 2011 at 11:05 AM, Vieri rentor...@yahoo.com wrote: Hi, I have a new Digium TE205P 2-span E1 card I just installed on a server. As soon as I boot the machine, the card's leds flash red (ports 1 and 2) - even when in the BIOS. That's not good, right? I don't have another

Re: [asterisk-users] Best PBX for Call Centers?

On Thu, Dec 15, 2011 at 3:39 PM, Carlos Alvarez car...@televolve.com wrote: On Thu, Dec 15, 2011 at 11:33 AM, Tarek Sawah tareksa...@hotmail.com wrote: Hello List, I have customer with a 40 Agents call center. and is looking to install a PBX switch that can serve those agents. As per my

Re: [asterisk-users] What version to upgrade to...?

On Mon, Dec 12, 2011 at 12:26 PM, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Monday, December 12, 2011 8:27 AM To: Asterisk Users Mailing List -

Re: [asterisk-users] Walkie talkie to sip phone interface

On Wed, Nov 30, 2011 at 6:20 PM, Ferdinand Babas ba...@cfht.hawaii.edu wrote: Hi All, I've been trying to find a solution that would allow our sip phones to communication with walkie talkies.  Our setup is that we have sip phones setup in 2 locations, headquarters and dome.  We can

Re: [asterisk-users] Limit monthly calls by context

On Wed, Nov 16, 2011 at 10:46 AM, Hans Goossen goos...@planet.com.py wrote: Hello group, I have this situation: I have several contexts with a few extensions each one. I need to give every context a limited quantity of minutes they can use. All the extensions in the context will share the

Re: [asterisk-users] Issues with Xorcom astribanks when running Ubuntu 11.10 (oneiric) as a quemu/libvirt guest

On Sat, Oct 29, 2011 at 3:14 PM, Eric van der Vlist v...@dyomedea.com wrote: Hi, Xorcom astribanks get initialized straight on when using Ubuntu 11.10 packages but I am having a hard time to get the same result running in a qemu/libvirt image. The first difficulty is that astribanks devices

Re: [asterisk-users] nvfaxdetect in 10.0

On Tue, Oct 18, 2011 at 6:21 PM, Danny Nicholas da...@debsinc.com wrote: Hi gang,     We are moving our 1.4 asterisk with DAHDI over to 10.0 with SIP.  Everything is going nicely except that I can’t get NV_FAXDETECT to compile properly into 10.0.  Because of this, I will have to

Re: [asterisk-users] asterisk hardware

On Tue, Oct 4, 2011 at 6:06 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Fri, Sep 30, 2011 at 10:50:12AM -0400, Adam Moffett wrote:  Is there any reason not to run Asterisk on an Intel Atom board? Only if it's not strong enough. Note that Atom may mean some different things. So

Re: [asterisk-users] Beep file with Record

On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion arjan.kr...@mobillion.nl wrote: This is my complete CLI logging -- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37, /var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60) in new stack [Oct  4

Re: [asterisk-users] Overlap SIP dialing

On Thu, Sep 8, 2011 at 9:38 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 09/07/2011 11:06 AM, Daniel Tryba wrote: The aim of the quest for overlap dialing is to let the user enter a number at their own pace but immediatly dial when all digits are received (just like plain old ISDN

Re: [asterisk-users] Overlap SIP dialing

On Thu, Sep 8, 2011 at 11:21 AM, Olle E. Johansson o...@edvina.net wrote: 8 sep 2011 kl. 17:17 skrev Kevin P. Fleming: Honestly, I'm not really sure that there is a practical solution here. ISDN overlap dialing was intended for 'dumb' phones, and SIP phones aren't 'dumb' :-) That's a

Re: [asterisk-users] Overlap SIP dialing

On Wednesday, September 7, 2011, Olle E. Johansson wrote: 7 sep 2011 kl. 15:59 skrev Daniel Tryba: Looking at the history of the list I don't expect any answer but lets try anyway: Does anybody use overlap dialing from SIP devices to asterisk? Does anybody have a working example?

Re: [asterisk-users] Overlap SIP dialing

On Wed, Sep 7, 2011 at 10:28 AM, Olle E. Johansson o...@edvina.net wrote: 7 sep 2011 kl. 16:20 skrev Andrew Latham: On Wednesday, September 7, 2011, Olle E. Johansson wrote: 7 sep 2011 kl. 15:59 skrev Daniel Tryba: Looking at the history of the list I don't expect any answer but lets

Re: [asterisk-users] USB or Ethernet based FXO device ?

On Wed, Aug 31, 2011 at 11:49 AM, Carlos Chavez cur...@telecomabmex.com wrote: On Wed, 2011-08-31 at 17:03 +0200, Marco Signorini wrote: Hi. I was following this thread. We normally use Patton SmartNode SN4112 series to interface to FXO ports. But I'm looking for something different for a

Re: [asterisk-users] Bind SIP over TCP port in asterisk 1.4.42.

On Thu, Aug 25, 2011 at 9:26 AM, Catalin S. jonsonpla...@gmail.com wrote: Hello, I need to listen on tcp 5060 on my actual asterisk 1.4.42. I tried in sip.conf at [general] section the following options: transport=tcp tcpenable=yes tcpbindaddr=0.0.0.0 but after all that changes i still not

Re: [asterisk-users] Polycoms rebooting themselves

On Wed, Aug 17, 2011 at 6:01 PM, Mike Diehl mdi...@diehlnet.com wrote: Hi all, I've got a customer with 10 Polycom 335's and the latest(ish) firmware.  For the most part, things are working well. However, about once a day, a given phone will just reboot.  They don't do it all at once, and

Re: [asterisk-users] Polycoms rebooting themselves

On Wed, Aug 17, 2011 at 6:35 PM, Mike Diehl mdi...@diehlnet.com wrote: On Wednesday 17 August 2011 4:12:21 pm Doug Lytle wrote: Mike Diehl wrote: Any other ideas? They should be writing out logs to your ftp server (If your provisioning them that way). At the moment, my web server isn't

Re: [asterisk-users] Where to proceed next

On Thu, Aug 11, 2011 at 3:04 PM, Danny Nicholas da...@debsinc.com wrote: Hello list,       I presently use the 1.4 releases because I enjoy sleeping at night.  I understand that 1.4 reaches end-of-life in a little over 8 months

Re: [asterisk-users] burned module X400M

On Thu, Aug 4, 2011 at 10:36 AM, Agustina Berretta agustina.berre...@gmail.com wrote: Hello folks! How can I be sure a module was burned by high tension? I installed the module, configured it using dahdi_genconf -vv but when I type: asterisk -rx dahdi show channels I don´t see the module.

Re: [asterisk-users] Lightning and thunder

On Wed, Jul 27, 2011 at 9:44 AM, Claude Hayn chayn...@gmail.com wrote: We are frequently losing power during lightning storms.  (Yes we have UPS, but often by the time power comes back up the UPS has run out of juice) We are using Asterisk with a T1/PRI card as a front end connected to our

Re: [asterisk-users] [1.4] Minimal installation?

On Mon, Jul 18, 2011 at 9:20 AM, Gilles codecompl...@free.fr wrote: Hello, I'd like to run Asterisk on an embedded device, where space is scarce. It should be able to handle calls from a VoIP provider in SIP, calls from the PSTN through Dahdi, and voicemail. If someone's already done this,

Re: [asterisk-users] Using Firewall to protect Asterisk

On Fri, Jul 15, 2011 at 12:47 PM, CDR vene...@gmail.com wrote: I need to keep out all connection from 5 countries, which originate most of the Denial of Service attacks. The entries are around 9000 if used as xx.xx.0.0/16. I heard that there is a smarter way to do this by using User Tables in

Re: [asterisk-users] TDM400p susceptible to EMI?

On Wed, Jul 13, 2011 at 2:51 PM, Steve Edwards asterisk@sedwards.com wrote: I have a TDM400p with 3 fxs and 1 fxo daughter cards. It's in a mini-itx case with a 'right-angle' PCI riser card so the TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5 hard drive. I'm getting a

Re: [asterisk-users] TDM400p susceptible to EMI?

riser card so the TDM400p is 'sandwiched' between the Atom D525 CPU and the 2.5 hard drive. I'm getting a bunch of clicks and pops on all ports. Has anybody had a similar experience? Did you find a solution? On Wed, 13 Jul 2011, Andrew Latham wrote: How is it grounded? Silly I know

Re: [asterisk-users] References customers

On Sun, Jul 10, 2011 at 5:22 PM, bilal ghayyad bilmar...@yahoo.com wrote: I mean: What are the customers (big customers I mean) that they installed Asterisk in their company to be as a reference? Example: Toyota, GM, Hilton, Shiraton hotel, ... etc An example of such companies, whom? Is

Re: [asterisk-users] How to without GUI

On Fri, Jul 1, 2011 at 1:55 PM, Danny Nicholas da...@debsinc.com wrote: Hey gang,     I’ve got a CISCO SPA3102 that I want to set up.  My environment is not favorable for using the Asterisk GUI interface – does anybody have step by step how to set up a SIP trunk just by

Re: [asterisk-users] How to without GUI

On Fri, Jul 1, 2011 at 2:23 PM, Doug Lytle supp...@drdos.info wrote: Danny Nicholas wrote: step by step how to set up a SIP trunk just by editing shudder sip.conf You'll find that most here don't use a GUI. Doug Doug Many people get addicted to the users.conf and res_phoneprov for

Re: [asterisk-users] Clarification of the terms shown on CLI

On Tue, Jun 28, 2011 at 4:53 PM, Bruce B bruceb...@gmail.com wrote: Hi everyone, When doing a sip show settings on Asterisk 1.6.2.18, I see the following:   Match Auth Username:    No   Allow unknown access:   Yes   Allow subscriptions:    Yes   Allow overlap dialing:  Yes   Allow promsic.

Re: [asterisk-users] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko

On Mon, Jun 20, 2011 at 11:47 PM, Alex Balashov abalas...@evaristesys.com wrote: I nominate this for most imaginative use of Asterisk-users of 2011. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax:

Re: [asterisk-users] Dahdi 2.4.0 and Squeeze

On Tue, Jun 14, 2011 at 9:44 AM, Olivier oza_4...@yahoo.fr wrote: Hi, I'm using a two-years old installation script for the first time on a Squeeze (linux 2.6.32) platform. For an unknown reason (might be an obvious one), Dahdi can't be loaded anymore. 1. First of all, it seems /dev/dahdi

Re: [asterisk-users] TFTP to be installed in Linux same asterisk machine to be used with Cisco

On Sat, Jun 11, 2011 at 10:29 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; Any one can suggest a TFTP server to be installed in Fedora (same machine that Asterisk is installed) to be used for Cisco IP Phones to download the required firmware and configuration files. Thanks for the

Re: [asterisk-users] issues.asterisk.org/jira not working

On Wed, Jun 8, 2011 at 3:20 PM, Russell Bryant russ...@digium.com wrote: A number of people are reporting that Safari is not working properly with JIRA.  Use Firefox or Chrome for now. -- Russell Bryant Digium, Inc.   |   Engineering Manager, Open Source Software 445 Jan Davis Drive NW    

Re: [asterisk-users] issues.asterisk.org/jira not working

On Wed, Jun 8, 2011 at 5:56 PM, Satish Patel satish...@hotmail.com wrote:  It not working on iPhone. It's saying not able to make secure connection -- Sent from my iPhone Satish, Can you share what the SSL/TLS Cert says? Safari and mobile platforms have a smaller list of CAs, just to make

Re: [asterisk-users] OT, free software for SIP ladder diagrams?

On Tue, May 17, 2011 at 1:12 PM, Steve Edwards asterisk@sedwards.com wrote: I was debugging a turnup with Global Crossing the other day and they presented me with a web page that displayed a 'ladder diagram' of a call including a ton of detail all neatly organized in tabs and links so you

Re: [asterisk-users] OT, free software for SIP ladder diagrams?

, 17 May 2011, Andrew Latham wrote: I think there are some, I saw one mentioned on the BACNet mailing list...  ahh yes http://cloudshark.org/ takes tshark and wireshark uploads... close to what you are looking for... Thanks, but not unless I've missed a lot on first glance. Cloudshark looks

Re: [asterisk-users] Light indicator managed by Asterisk

On Thu, May 12, 2011 at 12:50 PM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like

Re: [asterisk-users] Really, really loud ringers

On Mon, May 9, 2011 at 4:40 PM, Justin Sherrill justin.sherr...@americanrocksalt.com wrote: Anyone have some recommended equipment for alerting people to calls in a noisy environment? I have Polycom IP550 phones set up in some really noisy environments - our mine hoists - and they tend to

Re: [asterisk-users] Supermicro X7SPE (Atom) as Asterisk server?

On Sat, May 7, 2011 at 3:05 AM, Vahan Yerkanian va...@arminco.com wrote: On 5/6/11 11:52 PM, Andrew Latham wrote: On Fri, May 6, 2011 at 2:48 PM, Vahan Yerkanianva...@arminco.com  wrote: Has anyone used this board as an Asterisk server? http://www.supermicro.com/products/motherboard/ATOM

Re: [asterisk-users] Supermicro X7SPE (Atom) as Asterisk server?

On Sat, May 7, 2011 at 10:12 AM, Ryan Wagoner rswago...@gmail.com wrote: On Fri, May 6, 2011 at 2:52 PM, Andrew Latham lath...@gmail.com wrote: On Fri, May 6, 2011 at 2:48 PM, Vahan Yerkanian va...@arminco.com wrote: Has anyone used this board as an Asterisk server? http://www.supermicro.com

Re: [asterisk-users] Supermicro X7SPE (Atom) as Asterisk server?

On Fri, May 6, 2011 at 2:48 PM, Vahan Yerkanian va...@arminco.com wrote: Has anyone used this board as an Asterisk server? http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=HIPMI=Y I'm mostly interested about the possible compatibility issues this board may have with the

Re: [asterisk-users] receive faxes

On Wed, May 4, 2011 at 10:12 AM, Eric Wieling ewiel...@nyigc.com wrote: Does Hylafax support T.38? The free fax works just fine with DAHDI.  I've never tried to do T.38 with that since it seems like it would be complicated and not give me much over using DAHDI. There is the t38modem[1]

Re: [asterisk-users] receive faxes

On Wed, May 4, 2011 at 12:00 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: ... (For my part, I'm actually surprised that nobody came up with a proper protocol for encapsulating the stream of zeros and ones that make up a fax transmission but rely on the precise timing inherent with a

Re: [asterisk-users] Having redundancy, so if first IP failed then send for the other

On Tue, May 3, 2011 at 5:31 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; I need to configure the SIP account so if first IP address failed then to send for the second IP address. How to do this? While configuring the sip account, at the host parameter, can I give two IP addresses

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

On Thu, Apr 28, 2011 at 1:34 PM, satish patel satish...@hotmail.com wrote: Where did you download asterisk 1.10 or trunk ? I search and found nothing. could your point me there? -S svn co http://svn.asterisk.org/svn/asterisk/trunk /usr/src/asterisk_trunk -- ~~~ Andrew lathama Latham

Re: [asterisk-users] DHCP / DNS

No [1] 1. AsteriskNOW does have some of these services as do many distributions like Zentyal. On Wed, Apr 27, 2011 at 2:04 PM, Thomas Perron thomas.per...@gmail.com wrote: Are there any internal DHCP or DNS services built-in to the Asterisk code? -- --

Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?

On Wed, Apr 27, 2011 at 3:34 PM, Olle E. Johansson o...@edvina.net wrote: Friends, We have a discussion on asterisk-dev about the maintenance of the 1.4 branch. According to the release plans, support for 1.4 was scheduled to close in April 2011 - basically now. After that, only security

Re: [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!

On Mon, Apr 25, 2011 at 11:10 AM, Paul Belanger pabelan...@digium.com wrote: On 11-04-25 10:49 AM, David Backeberg wrote: On Mon, Apr 25, 2011 at 10:40 AM, C. Savinovich c.savinov...@itntelecom.com  wrote: Does this ConfBridge requires a hardware timing source? No, and neither does MeetMe

Re: [asterisk-users] core show channels consise in asterisk 1.8.3

On Mon, Apr 18, 2011 at 3:06 PM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 11-04-18 02:47 PM, Jerry Geis wrote: When I do core show channels concise over the AMI interface how do I specify that I want to see the actual channel number like DAHDI/4/xxx where 4 is the actual

Re: [asterisk-users] Asterisk thread limit

On Wed, Apr 13, 2011 at 10:50 AM, satish patel satish...@hotmail.com wrote: Hi Guys! I'm middle of testing my asterisk-1.8.3.2 just make sure how much call it could handle in production so following is my senario. [sipp_client]---[Asterisk][sipp_server]

Re: [asterisk-users] asterisk and fail2ban

On Wed, Mar 30, 2011 at 9:38 AM, vip killa vipki...@gmail.com wrote: so does anyone use fail2ban w/ asterisk or most people use sshguard? Vip, the overall message is that it takes layers of settings/configurations to secure an installation. Simple Guide 1. alwaysauthreject = yes in

Re: [asterisk-users] Debugging not going to log file

On Tue, Mar 29, 2011 at 11:05 AM, Dean Hoover kb7...@gmail.com wrote: I have an Asterisk server running 1.6.2.13, where I can't seem to get the increased logging to save to the /var/log/asterisk/messages file. I have tried using the standard core set debug 10 and core set verbose 10, as well

Re: [asterisk-users] asterisk and fail2ban

On Tue, Mar 29, 2011 at 2:34 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On 3/29/2011 12:25 PM, Steve Edwards wrote: On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan First thing I'd do is restrict the ip blocks your sip endpoints can register/call from in sip.conf (or your

Re: [asterisk-users] asterisk and fail2ban

On Mon, Mar 28, 2011 at 9:20 AM, vip killa vipki...@gmail.com wrote: Is anyone using asterisk with fail2ban? I have it working except it takes way more break-in attempts than what is set in maxretry in jail.conf For example, I get an email saying: The IP 199.204.45.19 has just been banned by

Re: [asterisk-users] White papers or success cases to convince a customer?

On Fri, Mar 25, 2011 at 7:05 PM, Carlos Chavez cur...@telecomabmex.com wrote:     Can anyone recommend some White Papers or Success Cases that we can use to ease the mind of a customer that has not heard much about Asterisk?  All they know is Avaya at this point. -- Carlos Chavez Director

Re: [asterisk-users] One PRI card with 2 (or more) Telcos

On Fri, Mar 18, 2011 at 3:15 PM, Tiago Geada tiago.ge...@gmail.com wrote: Just a follow up with a bit more information asterisk*CLI module show like timing Module                         Description                              Use Count res_timing_pthread.so          pthread Timing

Re: [asterisk-users] One PRI card with 2 (or more) Telcos

On Fri, Mar 18, 2011 at 3:17 PM, Adrian Serafini adrian-li...@wombit.com wrote: Is there a problem having 2 telcos on the same PRI card? I think you go with one master timer as the Telco.  Then the other spans are secondary, tertiary, quaternary timers. Adrian Adrian This only works when

Re: [asterisk-users] Extract Remote-Party-ID from incoming INVITE indialplan

On Wed, Mar 16, 2011 at 10:50 AM, Jonas Kellens jonas.kell...@telenet.be wrote: is it possible to extract the Remote-Party-ID from an incoming call in the dialplan ? Is there some kind of function for this ? Kind regards, Jonas. 1.8 Documentation on Connected Line update. Works like magic.

Re: [asterisk-users] Is asterisk 1.8 stable version to upgrade from asterisk 1.6 on live server?

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions -- ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] doorphone?

On Wed, Mar 9, 2011 at 12:53 PM, Darrick Hartman (lists) dhart...@djhsolutions.com wrote: On 03/09/2011 02:57 AM, Dan Journo wrote:   could anybody suggest a usable doorphone and magnetic door opener   hardphone system for me, please? Of course should be connectable to   asterisk. I am in

Re: [asterisk-users] server performance....

On Fri, Mar 4, 2011 at 2:13 PM, viswavardhanreddy karna viswavardhanre...@gmail.com wrote: Hi every one, I am doing some experiments on asterisk server performance.. How can we know server performance? can any one explain me plz  I have 2 doubts regarding the

Re: [asterisk-users] Recieve_Fax caused crash 1.8.2.3

On Fri, Feb 25, 2011 at 9:49 AM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 11-02-24 08:56 PM, Andrew Latham wrote: And I go back to triple check and compare revision numbers...  You are 100% correct, the revision numbers in our local repository are wrong, someone pushed the 1.8.3 RC3

Re: [asterisk-users] Recieve_Fax caused crash 1.8.2.3

On Thu, Feb 24, 2011 at 6:05 PM, Bryant Zimmerman brya...@zktech.com wrote: I had an issue today where receive_fax caused an asterisk switch to crash. The switch is 1.8.2.3 version. The call was coming from a fax machine. The call started receive_fax answered and then asterisk stopped

Re: [asterisk-users] Recieve_Fax caused crash 1.8.2.3

On Thu, Feb 24, 2011 at 7:43 PM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 11-02-24 04:08 PM, Andrew Latham wrote: There are many updates in 1.8.2.4 that may fix your issue.  If you are running any version of 1.8 it should be a quick update. I wouldn't say many. There is one fix

Re: [asterisk-users] Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available

On Tue, Feb 22, 2011 at 12:16 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Has this issue been fixed in this release of 1.8 (or even in the previous 1.8.2.3)? https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 Thanks Ish Ishfaq, I spoke to soon and was looking at the

Re: [asterisk-users] Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available

On Tue, Feb 22, 2011 at 12:16 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Has this issue been fixed in this release of 1.8 (or even in the previous 1.8.2.3)? https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 Thanks Ish snip -- Ishfaq Malik Software Developer

Re: [asterisk-users] Cisco 7945G phone with asterisk

On Wed, Feb 16, 2011 at 7:32 AM, ast guy ast...@gmail.com wrote: Hi,  Anyone who has deployed Cisco 7945G phone with asterisk, kindly share your experience. /ag I did some 7941's a few months ago with SIP. They work pretty well. Make a console cable for the AUX port and you can see them

Re: [asterisk-users] Connect Asterisk to a cell phone

On Wed, Feb 16, 2011 at 2:49 AM, logan logan...@gmail.com wrote: Hello, Are there any gateways which allow me to hook a cellphone to Asterisk and use that line for routing my calls? Basically, I'm looking to play around a bit and if I can get to connect a cellphone with Asterisk then that

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