Re: [asterisk-users] shell dialplan application blocking

Use AGI Kind regards, Matt > On Jun 4, 2018, at 02:28, Benjamin Marty wrote: > > I'm calling a script which needs to wait a certain time and also hold the > call for this time. But the script dialplan application seems to work non > blocking. Is there a way to hold the call/dialplan till

Re: [asterisk-users] Asterisk chan_sip registration attempts

Maybe the provider has added an extra gateway and it is not processing accounts correctly. If they had one before and now two then 40-60% registration fails would show that. Kind regards, Matt > On Oct 10, 2017, at 06:27, Dmitriy Ermakov wrote: > > Hello! > > Could

Re: [asterisk-users] softphone instead of desktop phones

I use Bria on all of the above. Kind regards, Matt > On Apr 29, 2017, at 10:35 AM, Thomas wrote: > > Hello, > Iam lookong for an Softphone for iPhor oder Android smartphone using togehter > with an headset. > I tried Zoiper and CSipSimple but quality was bad compared

Re: [asterisk-users] Call List Campaign to an IVR

Not really, doing the way below you don't even have to worry about it. They both go out at the same instant and as soon as it hits voicemail it disconnects the other leg. If you wanted you could leave it ringing for twenty minutes and it would still have the same effect. Kind regards, Matt

Re: [asterisk-users] Surfing the web via Asterisk.

This is a really interesting project but I think it's going to be seriously hard. You're going to need to parse meaning from a site, and that's not an easy thing to do. If you're focused on a few of the bigger sites then it might be easier. You almost want a middle layer that can parse

Re: [asterisk-users] Installing Asterisk on MAC native

It pretty much just works the same way as Linux. you might need to use brew to install a few prerequisites but I've got it running on my MacBook Pro without any major problems. It's good for testing things but I wouldn't use a MacBook as an office server or anything. And to be fair most of

Re: [asterisk-users] 1000 analogue lines with asterisk

There is definitely no way you should put 1000 lines on a single box. To be honest I do wonder what you want to do with 1000 lines as your description probably changes the recommendations. Kind regards, Matt > On Feb 17, 2016, at 5:09 PM, Goke Aruna wrote: > > Thanks

[asterisk-users] No sound with internal calls depending on which phones

Dear all, I have a very strange problem : * external calls work perfectly, * internal calls between some phones too, * but internal call between two similar phones don't work !!! (Snom 710) When we have sound, there are no errors in asterisk. When we do not have sound, there is the

Re: [asterisk-users] No sound with internal calls depending on which phones

[asterisk-users] No sound with internal calls depending on which phones You might have to disable srtp negotiations inside the phone web ui options. Mitul On Nov 12, 2015 8:53 PM, "(lists) Denis BUCHER" <dbuche...@hsolutions.ch <mailto:dbuche...@hsolutions.ch>>

Re: [asterisk-users] Looking for PRI Card with automatic fail over

There was a product called something like red box or similar that I saw around 5 years ago. Probably not entirely helpful but maybe Google will help. Kind regards, Matt On Aug 3, 2015, at 9:50 AM, Eric Klein eric.kl...@greenfieldtech.net wrote: Hi all, Strange request, I have a

Re: [asterisk-users] How to use TRUNK only if IAX fails?

The command he gave you was in Asterisk. Why do you not want to call it to try it? Then you can fail over to the other trunk if the IAX link is down. Kind regards, Matt On May 30, 2015, at 2:03 AM, Ashwin Surendran ashwin.surend...@now-health.com wrote: Many Thanks Carlos, I was hoping

[asterisk-users] Asterisk call forward for T1 incoming calls

Is there a way to divert incoming calls on DAHDI T1 channels so telco gets the diversion and send the call to new number and releasing the channel? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

[asterisk-users] handset forwarding Diversion header cannot be set on Local channels

is there anyway to change Sip headers in local channels? if a user sets forward on their handset, calls coming in to the handset get diversion header added: Diversion: 202 sip:202@192.168.1.46;reason=deflection Then asterisk sends the call to local channel: - Now forwarding SIP/201-0483 to

Re: [asterisk-users] AppVoicemail overwrites voicemail.conf

yes, thanks you! On Sat, Mar 22, 2014 at 9:13 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Fri, Mar 21, 2014 at 11:58 PM, Al lists asteris...@gmail.com wrote: looking more into this, looks like this is not a issue, its related to users changing voicemail password from

[asterisk-users] AppVoicemail overwrites voicemail.conf

We noticed issues with voicemail and somehow looks like voicemail.conf has been overwritten: ;! ;! Automatically generated configuration file ;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf) ;! Generator: AppVoicemail ;! Creation Date: Thu Mar 20 06:48:16 2014 ;! i saw a bug for 1.4

Re: [asterisk-users] AppVoicemail overwrites voicemail.conf

...@polybeacon.com wrote: On Fri, Mar 21, 2014 at 3:22 PM, Al lists asteris...@gmail.com wrote: We noticed issues with voicemail and somehow looks like voicemail.conf has been overwritten: ;! ;! Automatically generated configuration file ;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf

Re: [asterisk-users] AppVoicemail overwrites voicemail.conf

looking more into this, looks like this is not a issue, its related to users changing voicemail password from handset, asterisk rewrites the file. On Fri, Mar 21, 2014 at 9:31 PM, Al lists asteris...@gmail.com wrote: passwordlocatio seems to be related to vmsecret from voicemail.conf sample

Re: [asterisk-users] mixmonitor extension

On 24-01-14 00:37, Marek Cervenka wrote: can someone confirm that mp3 is unsupported? is patch available? Iirc mp3 was in asterisk-addons in asterisk 1.4 and iirc in later versions of asterisk you can enable format_mp3 in make menuselect. what about patch for Opus? uncle google doesnt

Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is it stable?

On 16-01-14 21:37, Gergely Kiss wrote: Dear List, I'm about to build an Asterisk 11.7 based PBX from scratch for our company. I'm in the middle of the planning phase and it turned out that our VoIP provider prefers H.323 protocol for handling voice calls (while SIP is also supported as plan B).

Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is it stable?

On 17-01-14 01:57, Dan Austin wrote: Patrick Lists wrote: On 16-01-14 21:37, Gergely Kiss wrote: Dear List, I'm about to build an Asterisk 11.7 based PBX from scratch for our company. I'm in the middle of the planning phase and it turned out that our VoIP provider prefers H.323 protocol

Re: [asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?

Hi Steve, On 15-01-14 18:53, Steve Edwards wrote: On Wed, 15 Jan 2014, Patrick Lists wrote: Would you mind sharing where you get the per country IP ranges from? I confess I 'brute forced' it by entering '/8s' into ARIN's web page and noting if the block had been assigned to a 'foreign' NIC

Re: [asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?

Hi Steve, On 14-01-14 10:39, Steven Howes wrote: On 14 Jan 2014, at 02:19, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: Thanks for your feedback Paul. The not having outbound trunks is going to be a challenge. Why? it’s what contexts were invented for. Yes that is indeed what

Re: [asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?

Hi Steve, On 15-01-14 02:44, Steve Edwards wrote: On Tue, 14 Jan 2014, Patrick Lists wrote: ...I guess I'll cook up some dialplan logic that records IP addresses, keeps track of the amount of failed password attempts etc. and block the offending IP addresses... A few iptables rules can

[asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?

Hi all, I'm looking into adding the ability to call me at m...@mydomain.org on my Asterisk 11 box. Does anyone have any tips or dialplan snippets to allow this kind of access as securely as possible? Thanks, Patrick -- _ --

Re: [asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?

On 14-01-14 02:36, Paul Belanger wrote: On Mon, Jan 13, 2014 at 9:24 AM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: Hi all, I'm looking into adding the ability to call me at m...@mydomain.org on my Asterisk 11 box. Does anyone have any tips or dialplan snippets to allow this kind

[asterisk-users] is this expected behaviour?

i noticed in asterisk 10.12.3, i get messages like this: [2014-01-08 19:03:59] NOTICE[2987]: chan_sip.c:23900 handle_request_invite: Failed to authenticate device 305sip:3...@my.server.ip;tag=0d516e63 but not mentioning attacker ip (to be used for fail2ban) is this expected? --

Re: [asterisk-users] Problem building dahdi from source

On 01/03/2014 03:56 PM, Jonas Kellens wrote: Hello, I am getting the following error when compiling dahdi : [snip] `/usr/src/dahdi-linux-complete-2.7.0.1+2.7.0.1/linux' make: *** [all] Error 2 I have the right kernel sources installed : [root@sip dahdi-linux-complete-2.7.0.1+2.7.0.1]#

Re: [asterisk-users] Why doesn't Asterisk try to prevent transcoding

On 12/15/2013 09:55 PM, CDR wrote: I have had the issue for years. The problem is that Asterisk developers are removed from the business. We desperately need simple way to eliminate transcoding when unnecessary. Transcoding brings a server to its knees. It is a very simple new setting in

Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk

On 12/14/2013 01:29 AM, Martin wrote: If I need to use SIP, from where to get the suitable firmware for these Cisco IP Phones 7942G? Be careful, not all versions of SIP firmware work with asterisk. I do have 8-3-1 (cmterm-7941_7961-sip.8-3-1)here and it works just fine with my 7961.

Re: [asterisk-users] Asterisk on Windows

Probably feeding the trolls but here it goes. On 12/04/2013 04:19 PM, CDR wrote: Digium is 100% lost in the map. If they would come up with a Paid version of Asterisk, one that would use the .NET framework in Windows, something simple to install, they could go public on the product. IIRC

Re: [asterisk-users] dahdi-tools 2.8.0-rc4 - udev rules not installed?

On 12/03/2013 06:35 PM, Russ Meyerriecks wrote: This is why we love release candidate feedback! Thanks! I've managed to mis-tag rc4 and missed all of Oron's commits. Cutting a v2.7.0.2 and a (correct) v2.8.0-rc5 today. Thanks. I'll give rc5 a spin when it arrives and report back if I find

Re: [asterisk-users] Problems compiling dahdi modules

On 12/02/2013 04:19 PM, Russ Meyerriecks wrote: This is fixed on the dahdi-linux master branch and will be included in the next release: More info: http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=summary

Re: [asterisk-users] DAHDI 2.7.0.1 and CentOS 6.5

On 12/02/2013 10:09 PM, Bakko wrote: Hello, during DAHDI 2.7.0.1 compilation on CentOS 6.5 64bit, I have this error: [snip] This was discussed earlier today and Russ pointed to the fixes: http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=summary

[asterisk-users] dahdi-tools 2.8.0-rc4 - udev rules not installed?

Hi, I just looked at 2.8.0-rc4 and noticed the udev rules/apps change which are now supposed to be part dahdi-tools. After make, make install and make config it seems the dahdi.rules are not installed. I couldn't find a reference to it in the Makefile either. Did I miss something or has the move

Re: [asterisk-users] Voicemail greeting playback issues?

On 11/26/2013 12:24 AM, Doug Lytle wrote: Bryant Zimmerman wrote: Hey all I believe I found the bug in Asterisk 11.xxx If someone can help me verify it. Actually, I wouldn't consider it a bug. I've know for years that you need to answer a channel before you play back audio or strange

Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

On 10/28/2013 07:29 PM, Eddie Mikell wrote: All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk

Re: [asterisk-users] Asterisk TON number

On 09/25/2013 09:22 AM, Endri Stefani wrote: Hi Greeting to all you out there. I am new at asterisk, I have been working with PLMN platforms telecommunication for 5 years with NSN and Huawei. We have recently built an asterisk PBX with Trixbox and connected it to our MSS using Digium E1

Re: [asterisk-users] Asterisk TON number

On 09/25/2013 01:57 PM, Endri Stefani wrote: Hi Patrick, If I use latest stable asterisk will I be able to change dialplan by changing pridialplan in chan_dahdi.conf? AFAIK yes. You may also want to check out Asterisk The Definitive Guide (4th edition is the latest). Paperback version:

Re: [asterisk-users] Somewhat-OT: Stupid NAT tricks to learn from Apple?

Hi Kristian, On 09/20/2013 03:17 PM, Kristian Kielhofner wrote: I've been spending some time looking at some of the significant changes Apple has made to Facetime in iOS 7. I'm far from an Apple fanboy but some of them are pretty interesting: - multiplexing everything over a single UDP port -

Re: [asterisk-users] ISDN outgoing caller id

On 08/27/2013 08:04 PM, Gergo Csibra wrote: Hi, is anybody out there who can set the outgoing caller id on ISDN (CAPI or misdn) channels? I've tryed everything what I found in forums, os voip-info.com but no luck. I use a fritz card with CAPI in my first installation (1 BRI), and a hfc 4 port

Re: [asterisk-users] Am I being hacked?

On 08/19/2013 08:10 PM, Eric Wieling wrote: One of Asterisk's dirty little secrets is that it does not show the source IP when a device or hacker tries sending a call without registering. The rejection message in the logs do not show the IP of the attacker. Yes it sucks, yes it has been

Re: [asterisk-users] Am I being hacked?

On 08/19/2013 08:55 PM, Steve Edwards wrote: On Mon, 19 Aug 2013, Ira wrote: [2013-08-18 05:56:29] NOTICE[17089][C-00a8] chan_sip.c: Failed to authenticate device 390sip:3...@xx.xx.xxx.xxx;tag=2762c06e xx.xx.xxx.xxx is my public I.P. What kind of filtering are you doing?

Re: [asterisk-users] Am I being hacked?

On 08/19/2013 09:29 PM, Eric Wieling wrote: Actually, you can try enabling the security logging destination in logger.conf. I believe that may contain the info, but it is new in Asterisk 11. 1.8 and earlier does not have this. Thanks I'll give that a try. Regards, Patrick --

Re: [asterisk-users] Echo Cancellation

On 07/25/2013 11:51 AM, bilal ghayyad wrote: Hello; If our Digium Telephony Card does not support echo cancellation like (1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome the echo? Use the free OSLEC echo canceller software module or Digium's commercial HPEC echo

Re: [asterisk-users] LUA

On 07/18/2013 03:56 PM, jacob.e.mi...@l-3com.com wrote: I am attempting to setup my server to use Lua for the dialplan (extentions.lua), but I am unable to get the asterisk configure script to find the installation of Lua on my box. I have downloaded the Lua sources from the www.lua.org site,

Re: [asterisk-users] Adjusting confbridge call quality

On 07/10/2013 06:46 PM, Chris Gentle wrote: [snip] and then others can connect via SIP. For some reason, when the speaker says words with S's and F's, they almost sound distorted. Not quite static but you can tell the quality has been affected. May just be a side-effect of 8,000 Hz. Just

Re: [asterisk-users] Asterisk 11 security log, fail2ban, drive-by SIP attacks

On 07/08/2013 01:46 PM, Giles Coochey wrote: Just a note that I did a little work to extend FreePBX distro with some extra Fail2Ban which deals with some drive-by SIP registration attempts. My regex is poor to middling, but the steps detailed here: http://www.coochey.net/?p=61 manage to stop

Re: [asterisk-users] Is it possible to provision lock Aastra phones?

On 07/06/2013 03:35 PM, Bruce B wrote: Thanks Patrick. Do the encrypted config files safe guard against hard resets such as Web Recovery mode - aka holding down 1 # sign at startup? My main purpose is to lock the sets due to contract terms so I'd rather not see user steal the phone and break

Re: [asterisk-users] Is it possible to provision lock Aastra phones?

On 07/06/2013 08:15 AM, Bruce B wrote: Hi everyone; Is it possible to provision lock Aastra phones to provider so that no soft or hard reset can unlock them? Iirc you can use encrypted configs using an app called anacrypt and lock them down. The admin guide (3.2.2) has more details in

Re: [asterisk-users] Digium Analog card and Asterisk

On 07/04/2013 05:32 PM, 杨华杰 wrote: Hi I just bought some digium analog cards and I would like to build an IVR system for my customers. However I am googling and googling , I didn't find any blog and instruction for beginners like me. So I come here for help. Any tips or blogs will help.

Re: [asterisk-users] Why does it take several seconds to interpret DTMF-input ?

On 06/11/2013 04:44 PM, Jonas Kellens wrote: [snip] Ok thanks. Any idea how I can resolve this ? Even if there *can* be more than 1 digit, in case there is only 1 digit it should go faster. Would it help if they pressed for example 1 followed by the # key? If not then, as Eric mentioned,

Re: [asterisk-users] OC3/STM-1 Line Card

On 06/09/2013 06:35 PM, Nick Khamis wrote: Anyone? Sangoma has a multiplexer: http://www.sangoma.com/products/stm1mux-fiber-multiplexer/ Which you could then use with: http://www.sangoma.com/products/a116-16-span-t1e1j1-board/ And there is this card:

Re: [asterisk-users] Which dahdi/libpri combo for BRI/PtmP ?

On 06/06/2013 05:55 PM, Olivier wrote: Hi, I need to rebuild a system which has 4 BRI ports and is connected to Point-to-multiPoint lines, in a country where telco often drop lines for energy savings. I think the dropped D-channel issue should be handled by a very recent DAHDI. If there are

Re: [asterisk-users] MeetMe exit status?

On 06/03/2013 06:47 PM, Matthew Jordan wrote: On 06/02/2013 08:36 PM, Patrick Lists wrote: Hi, Does MeetMe in Asterisk 11.4 set some kind of exit status or a var so I know for example if a conf ended normally or if someone gave a wrong conf number or pin? Thanks, Patrick

[asterisk-users] Missing dahdi-firmware RPM in asterisk-current repo at http://packages.asterisk.org

Hi, The dahdi-firmware package seems to be missing in the asterisk-current repo on http://packages.asterisk.org -- Finished Dependency Resolution Error: Package: dahdi-linux-2.6.2-1_centos6.x86_64 (asterisk-current) Requires: dahdi-firmware Can this please be fixed. Thanks,

[asterisk-users] MeetMe exit status?

Hi, Does MeetMe in Asterisk 11.4 set some kind of exit status or a var so I know for example if a conf ended normally or if someone gave a wrong conf number or pin? Thanks, Patrick -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] dahdi driver not getting install

On 05/13/2013 01:14 PM, Salaheddine Elharit wrote: hi You can download a tarball of the release here: http://downloads.asterisk.org/pub/telephony/dahdi-linux/dahdi-linux-2.6.2-rc1.tar.gz At least give the link to the latest release which is not 2.6.2-rc1 but 2.6.3-rc1:

Re: [asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones

Hi Carlos, On 04/28/2013 10:56 PM, Carlos Alvarez wrote: We have a new customer with a lot of old phones like the 9133i. They won't register, and we see some very strange behavior with them. If the SIP peer exists, they simply fail silently, with no error in the CLI or the messages log.

Re: [asterisk-users] fax - sound/tone - dealing with SPAM

On 04/04/2013 09:54 PM, Joseph wrote: +1.7044972383 If that number is his actual number, maybe create a script that calls him 10 times an hour, every hour between 00:00 - 07:00am and plays screaming monkeys every time he picks up (or his voicemail kicks in). Regards, Patrick --

Re: [asterisk-users] TigerJet 320G Chip / TDM400 Chipset / DAHDI Support

On 04/03/2013 02:48 PM, Marshall Henderson wrote: Hi Tzafrir- I know where to find the DAHDI source, but I was more asking where to actually find which chipsets are supported within the source. Any thoughts? Have you checked the PCI IDs in the source? Regards, Patrick --

Re: [asterisk-users] TigerJet 320G Chip / TDM400 Chipset / DAHDI Support

On 04/03/2013 08:34 PM, Marshall Henderson wrote: Hi Patrick- Yes, I did find the list of PCI IDs (I think). Do these look right (from wctdm.c): static DEFINE_PCI_DEVICE_TABLE(wctdm_pci_tbl) = { { 0xe159, 0x0001, 0xa159, PCI_ANY_ID, 0, 0, (unsigned long) wctdm }, { 0xe159,

Re: [asterisk-users] ERROR: Unknown signalling method ss7

On 03/14/2013 11:04 PM, mohsen feyzzadeh wrote: Hi all I installed DAHDI Version - 2.6.1 DAHDI Tools Version - 2.6.1 libss7-trunk Asterisk 11.0.1 from source on Fedora 12 x86_64. In case the 12 in Fedora 12 was not a typo, you do realize that Fedora 12 has been end-of-line for years and has

Re: [asterisk-users] digium card and virualbox

On 03/11/2013 04:18 AM, bilal ghayyad wrote: I am not mixing. I need this for LAB testing. How? This PCI passthrough, how to enable it on virualbox? It's in the VirtualBox manual. Regards, Patrick -- _ -- Bandwidth and

Re: [asterisk-users] Laptop error

On 03/11/2013 12:53 PM, termo termosel wrote: Hi, I have Ubuntu and Asterisk 11.2.1 in a boot USB. When I put it in desktop computer, asterisk starts without problem but if I insert the same USB in a laptop computer Asterisk doesn't start. Is it possible because different microprocessors?

Re: [asterisk-users] Sending SMS from asterisk

On 03/11/2013 07:07 PM, Asghar Mohammad wrote: HI Bilal, i am using chan_mobile for call termination, you can use it but you need to tweak chan_mobile.c it is broken from a long time. let me know if you want give it a try. If you could send the patches you made to chan_mobile to this mailing

Re: [asterisk-users] Serviced Office operator panel

On 03/12/2013 12:07 AM, Andrew Yager wrote: Hi, I'm trying to find (with some desperation now) a decent web based or application based UI that integrates with an Asterisk based PBX and is designed for a Serviced Office environment. Key features we're looking for: Don't know if it covers your

Re: [asterisk-users] Where can get the latest manual our user guide

On 02/08/2013 06:35 AM, Ding Peng wrote: Hi, everybody, Where can I get the manual or user guide of latest asterisk version, 1.11.x? I want to know the syntax and usage of all the supported functions or something like that in the latest version. You can find one on the O'Reilly website.

Re: [asterisk-users] DECT Solution

On 01/24/2013 10:37 AM, Zyumbilev, Peter wrote: Hello, I need to setup system of aroud 60 DECT phones with asterisk. So far I found http://www.grandstream.com/index.php/products/ip-voice-telephony/enterprise-ip-phones/dp715_710 However is there some cheap base station(similar to GSM cell) so

Re: [asterisk-users] clicking sound with alaw codec

On 01/24/2013 09:44 PM, Richard Kenner wrote: [snip] When I use alaw, the path from Asterisk to the Alcatel is completely clean, but the other way has a set of clicks that kind of sound like old-fashioned audio noise. [snip] It's been ages since I experienced that but things to check that come

Re: [asterisk-users] clicking sound with alaw codec

On 01/24/2013 11:57 PM, Richard Kenner wrote: - jitterbuffer settings (try on/off) I added jbenable=yes and get lots of: [Jan 24 17:53:41] WARNING[12317]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-6c7 received frame with invalid timing info: has_timing_info=1, len=0,

Re: [asterisk-users] Details process to configure Asterisk in CENTOS

On 01/22/2013 08:54 AM, Sakharam Thorat wrote: Can anybody send me Detailed process to configure Asterisk in CENTOS ?? Detailed description highly appreciated. Start by reading the Asterisk book, check asterisk.org and Google around to see if your question has already been answered.

Re: [asterisk-users] Need Help

On 01/17/2013 09:05 PM, Joe Ruffolo 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. Afaik Trixbox is no longer maintained and their forum are hardly active anymore so it may be a bit of a challenge to get

Re: [asterisk-users] Top Posting

themselves, and solving complex problems. I've seen many tech lists die off when people stop trying to help themselves and ask intelligent questions. Good point Carlos and I share your feeling. On the Postfix mailing list, when someone asks a basic how do I ... question, inevitably the response

Re: [asterisk-users] Top Posting

On 01/02/2013 06:20 PM, Steve Totaro wrote: I became a list member way before any such rule and never had to click through and agree to these update ToS. I am grandfathered in. Just looked it up. I see my first post back in April 2003, yours in September 2003 and Jon in March 2003. Wow you

Re: [asterisk-users] Top Posting

On 01/02/2013 09:46 PM, jon pounder wrote: On 01/02/2013 03:22 PM, Patrick Lists wrote: On 01/02/2013 06:20 PM, Steve Totaro wrote: I became a list member way before any such rule and never had to click through and agree to these update ToS. I am grandfathered in. Just looked it up. I see

Re: [asterisk-users] Top Posting

On 12/30/2012 04:26 PM, Ron Wheeler wrote: I participate in a lot of lists and top posting is now the norm since people want to see quickly if the message is worth reading. Isn't it a bit of a stretch to extrapolate your experience with your lists to top posting being the norm? I am

Re: [asterisk-users] Disappearing Call Files / Two threads dealing with my call files

There's no priority in your call file. Sent from my iPhone On 29/11/2012, at 11:12 PM, Necati Demir nde...@demir.web.tr wrote: Hello, I noticed that when i move a call file to outgoing directory, two asterisk threads are dealing with it. ]# grep FAX_44731.call /var/log/asterisk/full.2

Re: [asterisk-users] Asterisk crashing when trying to start a Jabber session with ejabberd

On 11/13/2012 12:11 AM, Phil Reynolds wrote: [snip] It turns out to be a known issue: https://issues.asterisk.org/jira/browse/ASTERISK-19532 ... and can be fixed by applying the patch at: https://issues.asterisk.org/jira/secure/attachment/43441/xmpp_no_crash_with_ejabberd.patch I will file

Re: [asterisk-users] Asterisk crashing when trying to start a Jabber session with ejabberd

On 11/13/2012 07:05 PM, Michael L. Young wrote: [snip] Is it an omission that this fix has not been applied to the 11 tree? From the looks of ASTERISK-19532 it seems that the fix has only been applied to 1.8 and 10. If you click on the link for ASTERISK-19532, there is a tab in the Activity

Re: [asterisk-users] B200p card - use dahdi or mISDN?

On 10/16/2012 08:50 AM, Sebastian Arcus wrote: I've just bought an OpenVOX B200p ISDN card - and if I remember correctly from last time I used one of these - it is possible to use either DAHDI or mISDN with it. Are there any factors to consider when choosing which software to use? Is one better

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

On 10/15/2012 09:07 AM, sudeep melekar wrote: [snip] i m completely new to asterisk so any help would be appreciated If you are totally new to Asterisk I recommend you first read the Asterisk book and go through the wiki. Both have sections how to install the various Asterisk components.

Re: [asterisk-users] Connecting Skype to Asterisk

On 10/12/2012 11:17 PM, Philip Bennefall wrote: From what I gather, it costs extra for each channel even for direct Skype to Asterisk calls. Since my plan was to use this for business purposes, I'd need at least something like 30 channels which would be way out of my monthly budget

Re: [asterisk-users] LDAP Driver and VoiceMail

On 10/04/2012 10:00 PM, Phil Daws wrote: Hello: I am investigating the possibility of using LDAP for storing certain Asterisk configuration parameters. I have examined res_ldap.conf and see where mailbox can be defined from AstAccountMailbox but I do not see where the password can be stored

Re: [asterisk-users] CDR Unanswered calls

On 10/05/2012 11:51 AM, Shanavaz E A wrote: Hi, No replies until now. Some one please help... There must be some people who are using it... Thanks No idea but since Asterisk is making you money why don't you hire an experienced Asterisk consultant to get it resolved. Regards, Patrick --

Re: [asterisk-users] How to log caller IP address in the CDR?

On 10/05/2012 02:10 PM, Benoit Panizzon wrote: Hello We had this situation: Some bot-net did try to guess SIP logins and finally succeeded. The Asterisk Server was abused to call a large number of expensive destinations. I'm sorry to hear that. In the Asterisk source there is a doc that

Re: [asterisk-users] FAX via Asterisk

On 09/27/2012 08:15 AM, Shanavaz E A wrote: [snip] Patrick, can you please give the steps to configure fax with iaxmodem and hylafax. Is it free to use? It's been years since I set it up so I don't know exactly how to configure it anymore. But I do remember that I found some howto/docs via

Re: [asterisk-users] 100 outgoing calls - 10 at a time - /var/spool/asterisk/outgoing/

On 09/28/2012 03:01 AM, Patrick Archibald wrote: Hi, Is there a way to move 100 .call files in to /var/spool/asterisk/outgoing/ at once and have Asterisk call at maximum 10 at a time? Afaik that is not possible. Wouldn't it make more sense to move call files in batches of 10 to outgoing/?

Re: [asterisk-users] FAX via Asterisk

On 09/26/2012 05:53 PM, Mark Robinson wrote: Hello. I have asterisk 1.8.18 which connects to ISDN PRI. All phones are sip, Aastra 6757i. Everything works as expected. We also have a FAX machine. We need to be able to use that FAX machine to send or receive faxes. We are planning to have a

Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR

On 09/25/2012 11:18 PM, Logan Bibby wrote: MyISAM would be best, in my opinion. The features that cause the little bit of performance overhead in InnoDB wouldn't be necessary for CDR storage. Iirc InnoDB is ACID compliant so might be preferable if MyISAM is not. More information here:

Re: [asterisk-users] Digium AEX410, MTNL Mumbai Caller-ID problems

On 09/14/2012 05:26 AM, Raj Mathur (राज माथुर) wrote: [snip] Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express) Your DAHDI and Asterisk versions are old so for starters I would update everything to the latest

Re: [asterisk-users] kernel: dahdi: Master changed to TE2/0/2 --- Is a normal message

On 09/14/2012 10:25 AM, A J Stiles wrote: [snip] It could be nothing more than a dry solder joint on one of the RJ45s. For the sake of five minutes' work with a soldering iron, that's got to be worth eliminating. Wouldn't that void your warranty? I would take it up with Digium support and

Re: [asterisk-users] Grandstream VoIP phones

On 01-09-12 04:14, Vladimir Mikhelson wrote: [snip] * Ability to send host name or other CN not equal to the phone IP in TLS negotiation Afaik you usually put alternative CNs in SubjectAltName in the certificate. Have you tried that? Regards, Patrick --

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

On 08/30/2012 09:45 AM, Gopalakrishnan N wrote: Hi, I have installed libpri, Dahdi 2.6 and Asterisk 1.8.15, in normal host, I am not using any virtualbox, still i struck in loading the modules. Please do not top post. Install strace and then start asterisk with the command: # strace asterisk

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

On 27-08-12 08:25, Gopalakrishnan N wrote: This is really tuff working with OpenSuse. I am clueless how to sort our this. Maybe switch to a different distribution? I have used CentOS and RHEL for years without any problems and as far as I know both debian and ubuntu should work without

Re: [asterisk-users] can we install 10 PCI card on asterisk

On 27-08-12 14:08, DHAVAL INDRODIYA wrote: Hi All, i would like to know if anyone has done or having idea regarding PRI terminations in asterisk. i have a requirement where i need to support 80 PRI in one machine i have found a machine which have 10 PCI slots available now i am thinking of

Re: [asterisk-users] Understanding CHANNEL function values

On 25-08-12 14:31, Stefan at WPF wrote: Hello all, I need some help understand the values of the CHANNEL function, e.g. txploss // local packets loss rxploss // remote packets loss txjitter // local jitter rxjitter // remote jitter My main problem in understand is that a

Re: [asterisk-users] xmpp / sip

Hi Hans, On 24-08-12 10:13, Hans Witvliet wrote: Hi all, After making a nice demo-setup for one of our innivationmanagers, he came up with a completely different stratagy ;-( Well if you could create it then obviously it's no longer innovative so they had to come up with something else :-)

Re: [asterisk-users] quick questions on version 10

I thought this was discussed and it was going to be left in? Sent from my iPhone On 23/08/2012, at 2:30 PM, Jerry Geis ge...@pagestation.com wrote: The AMI action CoreShowChannels deprecated the CLI concise command because the output of the AMI action is extensible without breaking existing

Re: [asterisk-users] Asterisk 1.8 and 11

On 22-08-12 20:04, Giuseppe Longo wrote: Just a little questions, what's the difference between asterisk 1.8 and asterisk 11? Iirc you can check the ChangeLog in the Asterisk 11 sources. Regards, Patrick -- _ -- Bandwidth

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

On 14-08-12 08:29, Gopalakrishnan N wrote: If I change autoload=no then asterisk is starting, but post to that loading modules even chan_sip.so asterisk hangs. Its strange, only in OpenSuse I am facing this. In CentOS, Ubuntu its working fine, same Asterisk version with same hardware. Please

  1   2   3   4   5   6   7   8   9   10   >