Re: [asterisk-users] OT - Gigaset C470IP - How to access SMS settings

2010/9/15 asterisk asterisk > Olivier, > > You should find out the SMS tab in the handset but not in the web service. > Did you IP pone work? > > CK > Hi, My phone is working OK but there is no SMS menu showing, though you can see this menu all around the user manual. How did you set yours ? D

Re: [asterisk-users] Speech To Text on linux with asterisk

Hello i have tried to convert through sphinx as suggested by Nickolay i am not getting convert my simple audio file. i am having following error while i fire following command pocketsphinx_continuous -infile /usr/etc/ask-propertyid.WAV -samprate 8000 \ -hmm /usr/etcSpeechToText/Communicator_sem

[asterisk-users] Digest Username/auth name misma tch‏

Hi I'm sorry. I mailed the same question again. because, it cannot be yet solved. any ideas with asterisk? [Aug 20 14:40:12] WARNING[29315]: chan_sip.c:11806 check_auth: username mismatch, have , digest has a...@192.168.0.1[aug 20 14:40:12] NOTICE[29315]: chan_sip.c:20479 handle_request_

[asterisk-users] about yahoo messager work with asterisk

Hi Asterisk Gurus Does it possible let asterisk work with yahoo messager? To let Yahoo messager have a voice call with SIP client? I had find that Yahoo message voice call use SIP . But did not know how it authentication with yahoo sip server side. Any advise is welcome!

Re: [asterisk-users] OT - Gigaset C470IP - How to access SMS settings

Olivier, You should find out the SMS tab in the handset but not in the web service. Did you IP pone work? CK On Tue, Sep 14, 2010 at 2:27 PM, Olivier wrote: > Hi, > > With my Gigaset C470IP (with latest 02223 firmware), I can't find a way to > access SMS settings from web configuration app or

Re: [asterisk-users] How different is implementing Cisco based system than Asterisk based system?

On Tue, 2010-09-14 at 15:56 -0400, Zeeshan Zakaria wrote: > Hello list, > > Slightly off the list topic, but I hope I'll get some help here. > Somebody wants me to implement for his project a Cisco based VoIP > system. I told him that I specialize in Asterisk based systems, but he > is not even aw

Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?

Thanks guys. I wasn't able to collect enough SIP debug as the problem was resolved as I was testing different configuration for the trunk. Probably a change on the provider side. John Novack: Unfortunately, it seems that this list has a non-stop list of people who like to stir up things or try to

Re: [asterisk-users] conf checkout

On 09/14/2010 02:51 PM, Steve Edwards wrote: > On Tue, 14 Sep 2010, Danny Nicholas wrote: > >> I see that some posters today don’t do full (or any?) backups of their >> Asterisk systems/configuration. This may (sort of) help you. Since >> pretty much all Linux systems have some sort of PERL inst

Re: [asterisk-users] How different is implementing Cisco based system than Asterisk based system?

I'll keep this all in mind. I don't plan to become a Cisco expert over night. Flirts I'll try to make them use Asterisk. I don't know the details yet. But some of these big organizations don't even want to consider anything other than the proprietary systems. Zeeshan A Zakaria -- www.ilovetovoip.

Re: [asterisk-users] How different is implementing Cisco based system than Asterisk based system?

I also thought that they should get it from an official Cisco reseller if they wanted support. Maybe at this stage they themselves don't know what they want. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-14 4:14 PM, "Peder" wrote: My best advice would be “don’t do it, it will only cause

Re: [asterisk-users] How different is implementing Cisco based system than Asterisk based system?

On 9/14/2010 3:09 PM, David Backeberg wrote: > On Tue, Sep 14, 2010 at 3:56 PM, Zeeshan Zakaria wrote: >> Now I have no previous experience with Cisco systems and don't want to screw >> up anything. Are they much different than Asterisk based systems? I guess >> the underlying VoIP technology is

Re: [asterisk-users] How different is implementing Cisco based system than Asterisk based system?

On Tue, Sep 14, 2010 at 3:56 PM, Zeeshan Zakaria wrote: > Now I have no previous experience with Cisco systems and don't want to screw > up anything. Are they much different than Asterisk based systems? sometimes. Cisco supports "SIP", but depending on the product, asterisk inter-networking with

Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11

I have compiled 1.6.2.10 and for now it seems the spontaneous reboots have stopped. I cross my fingers. If it still happens, I will post some 100 lines of my debug log round the time of happening. Jonas. On 09/14/2010 09:36 PM, Danny Nicholas wrote: -Original Message- From: asteri

Re: [asterisk-users] How different is implementing Cisco based system than Asterisk based system?

My best advice would be "don't do it, it will only cause headaches". It is completely different than * with different terminology, design considerations, etc. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Tue

Re: [asterisk-users] How different is implementing Cisco based system than Asterisk based system?

On Tue, Sep 14, 2010 at 3:56 PM, Zeeshan Zakaria wrote: > Now I have no previous experience with Cisco systems and don't want to screw > up anything. Are they much different than Asterisk based systems? I guess > the underlying VoIP technology is the same for both the systems so it > shouldn't be

[asterisk-users] How different is implementing Cisco based system than Asterisk based system?

Hello list, Slightly off the list topic, but I hope I'll get some help here. Somebody wants me to implement for his project a Cisco based VoIP system. I told him that I specialize in Asterisk based systems, but he is not even aware of Asterisk. The requirement of project is such that chances are s

Re: [asterisk-users] conf checkout

On Tue, 14 Sep 2010, Danny Nicholas wrote: I see that some posters today don’t do full (or any?) backups of their Asterisk systems/configuration.  This may (sort of) help you.  Since pretty much all Linux systems have some sort of PERL installed, these two files will let you make a quick copy

Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, September 14, 2010 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11

Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11

- "Jonas Kellens" wrote: > also the same question for you : what info am I actually looking for ?! > > Because my debug log can be very overwhelming when verbosity is 9. Well FFS read the links posted for you. Nobody is asking for what flies across your screen at 'verbosity 9' of the A

Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11

On 09/14/2010 09:21 PM, Danny Nicholas wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Tuesday, September 14, 2010 2:05 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re:

Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, September 14, 2010 2:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Spontaneous reboot

Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11

On 09/14/2010 09:12 PM, Carlos Chavez wrote: > On Tue, 2010-09-14 at 20:27 +0200, Jonas Kellens wrote: > >> And again !! Without me doing anything !! >> >> PBX Core settings >> - >>Version: 1.6.2.11 >>Build Options: LOADABLE_MODULES >>

[asterisk-users] conf checkout

Hi gang, I see that some posters today don't do full (or any?) backups of their Asterisk systems/configuration. This may (sort of) help you. Since pretty much all Linux systems have some sort of PERL installed, these two files will let you make a quick copy of any configuration or o

Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, September 14, 2010 2:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11

Re: [asterisk-users] DTMF

I've managed to get 'info' working. Not sure why the others didnt want to work. Thanks for your reply. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introducto

Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11

On Tue, 2010-09-14 at 20:27 +0200, Jonas Kellens wrote: > And again !! Without me doing anything !! > > PBX Core settings > - > Version: 1.6.2.11 > Build Options: LOADABLE_MODULES > Maximum calls: Not set > Maximum open file h

Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11

Core verbose is 9 !! What info can I give to you ?! What info can I get from my debug log ?! I don't see anything remarkable... What am I looking for ?! Jonas. On 09/14/2010 08:50 PM, Paul Belanger wrote: On Tue, Sep 14, 2010 at 2:27 PM, Jonas Kellens wrote: And again !! Without me do

Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11

Can you then please tell me what kind of debugging I need to enable ?! On 09/14/2010 08:41 PM, Steve Howes wrote: On 14 Sep 2010, at 19:27, Jonas Kellens wrote: And again !! Without me doing anything !! Yea, you didn't even enable any kind of debugging or anything. Amazing.. S

Re: [asterisk-users] DTMF

On Tue, Sep 14, 2010 at 1:33 PM, Dan Journo wrote: > It seems ive broken my settings and now, asterisk isnt detecting my DTMF > tones. > Can you revert to a working copy of .config files? If not, I would suggest some sort of backup procedures. Post your sip.conf to this thread, also how are you

Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11

On Tue, Sep 14, 2010 at 2:27 PM, Jonas Kellens wrote: > And again !! Without me doing anything !! > http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) b

Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11

On 14 Sep 2010, at 19:27, Jonas Kellens wrote: > And again !! Without me doing anything !! Yea, you didn't even enable any kind of debugging or anything. Amazing.. S -- _ -- Bandwidth and Colocation Provided by http://www.a

Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11

And again !! Without me doing anything !! PBX Core settings - Version: 1.6.2.11 Build Options: LOADABLE_MODULES Maximum calls: Not set Maximum open file handles: Not set Verbosity: 25 Debug level:

Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11

Hello, this is a *REAL* problem !! I did not reload or restart but look at the following : PBX Core settings - Version: 1.6.2.11 Build Options: LOADABLE_MODULES Maximum calls: Not set Maximum open file handles: Not set

Re: [asterisk-users] Call Recording Questions

Is there any way to prevent the end user hearing the *1 key tones when the touch recording is activated? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

[asterisk-users] DTMF

Hi, It seems ive broken my settings and now, asterisk isnt detecting my DTMF tones. What kind of diagnostics can I do to work this out? I've set the extension in sip.conf to everything listed on this page but no result. I've also played around with the settings on the phone with no help either

Re: [asterisk-users] sip show channels

RWC also wont show me how long I have to wait before I can restart. And while RWC is executing, will idle lines be able to make a new call, or will they have to wait till the restart is complete? If they have to wait, then I prefer manually restarting asterisk when I see a moment of quiet time. -

Re: [asterisk-users] sip show channels

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Tuesday, September 14, 2010 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip show channels True, that is even b

Re: [asterisk-users] sip show channels

True, that is even better. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-14 12:49 PM, "Steve Howes" wrote: On 14 Sep 2010, at 17:32, Dan Journo wrote: > I'm trying to view a list of the active calls to see i... Don't?. 'core restart when convenient' will wait until there are no calls

Re: [asterisk-users] sip show channels

> You should be doing "core show channels", even if all of your channels are > indeed SIP. Thats great thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introd

Re: [asterisk-users] sip show channels

Do a 'show channels'. You can also do 'show channels concise' or 'show channels verbose' for more details. In any case, it'll show you number of active calls at the end of output. Now some may point out to prepend 'core' before issuing these commands. I prefer to be brief, and used to this shorter

Re: [asterisk-users] sip show channels

On 14 Sep 2010, at 17:32, Dan Journo wrote: > I'm trying to view a list of the active calls to see if I can restart > Asterisk. Don't?. 'core restart when convenient' will wait until there are no calls. S -- _ -- Bandwidth

Re: [asterisk-users] sip show channels

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Tuesday, September 14, 2010 11:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] sip show channels Hi, I'm trying to view a

Re: [asterisk-users] Random File Name

Or 1234 => { Verbose (ID is ${UNIQUEID}); }; :) Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-14 12:27 PM, "Danny Nicholas" wrote: *>From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Zeeshan Zakaria >*Sent:* Tuesday, Sep

[asterisk-users] sip show channels

Hi, I'm trying to view a list of the active calls to see if I can restart Asterisk. When I do 'sip show channels', I get a huge list like this (just a sample pasted):- 92.110.7.210 (None) 198827f2469 00102/0 0x0 (nothing)No Init: OPTIONS 92.110.7.210 (None)

Re: [asterisk-users] Random File Name

>From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria >Sent: Tuesday, September 14, 2010 11:15 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [asterisk-users] Random File Name >Can you post wha

Re: [asterisk-users] Random File Name

Can you post what are you doing to see UNIQUEID? And also what version of Asterisk you are using? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-14 12:11 PM, "Dan Journo" wrote: > ${UNIQUEID} is going to be realtivly unique certnely in the short term I dont understand something. When I do

Re: [asterisk-users] Random File Name

Actually, ignore my last email. Forgot to reload. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk

Re: [asterisk-users] Random File Name

> ${UNIQUEID} is going to be realtivly unique certnely in the short term I dont understand something. When I do ${UNIQUEID}, I get something like this:- "SIP/215.166.5.140-0bbf" Is this correct? Its not a valid file name. Thanks Dan --

Re: [asterisk-users] PostgreSQL is asterisk friendly with it?

I use Postgres always and it is wonderful. Never use mysql so if you want a real DB just use Postgres *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---

Re: [asterisk-users] can asterisk accept "anonymous" register ?

On Tue, Sep 14, 2010 at 5:59 AM, zhou tianjun wrote: > I want to know does the asterisk can realize > that. Or  I > have to write module for that function ? > No, you need to tell Asterisk what to do. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pab

Re: [asterisk-users] High volume BLF - Suggestions?

On 13 September 2010 19:12, Cassius Smith wrote: > Steve > I have 64 channels being monitored with an SPA962 with two SPA932 > sidecars. It works perfectly with Asterisk 1.6.2.9; my users are very > happy with this. Latest firmware is a must. > > HTH > Cassius Smith > Any chance you could send me

[asterisk-users] question on asterisk 1.8 meetme

Currently using 1.4.X and looking to JUMP to 1.8 was reading the docs and have a question. in 1.4 I could do: /usr/sbin/asterisk -rx "meetme" to see all the current meetme's. I dont see what this is now in 1.8? Thanks Jerry -- _

Re: [asterisk-users] Speech To Text on linux with asterisk

On Tue, Sep 14, 2010 at 1:41 AM, DHAVAL INDRODIYA wrote: > -> Call comes in > -> start recording > -> call remains for 30 minutes > -> stop recording > -> convert wav file audio to text. > > is this possible with lumenvox or any other engine. > Not realistically, because you need to define grammar

Re: [asterisk-users] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?

This might help to answer poster's question. It tells how the allow anonymous sip connections work in FreePBX, and shows the code. http://www.geekzone.co.nz/sbiddle/7183 -- Zeeshan On Sun, Sep 12, 2010 at 12:11 AM, Paul Belanger < paul.belan...@polybeacon.

Re: [asterisk-users] Random File Name

On Tue, 14 Sep 2010, Dan Journo wrote: > Im looking at using MixMonitor to record calls and I know that I need to > set the filename first. > > However, with the number of calls coming in, hard coding the filename > isnt an option. > > So I need to do something like this:- > > MixMonitor(RAND

Re: [asterisk-users] Random File Name

On Tue, 14 Sep 2010, Gareth Blades wrote: > ${UNIQUEID} is going to be realtivly unique certnely in the short term > unless the sysyem is restarted. I mormally combine it with the date & > time to make it truly unique. UNIQUEID is the concatenation of AST_SYSTEM_NAME, the number of seconds sin

Re: [asterisk-users] Random File Name

I use ${UNIQUEID} with ${CDR(accountcode)} and it works great. But this is when you have an accountcode for each user. As the last poster suggest, you can append it with date and time and it'll be truly unique and also help you keep track of the recording. Zeeshan A Zakaria -- www.ilovetovoip.com

[asterisk-users] SIP 800 Origination/Termination - International

Anyone have a good provider for International (US/Canada at least) 800 termination/origination? I have a customer that had us port one of their 800 numbers and apparently didn't realize that they had published that number in Canada as well. Our current origination/termination provider can't han

Re: [asterisk-users] Speech To Text on linux with asterisk

If it works even half decent, kindly post your result on the list. I need something similar for a client, that is voicemail to text. After my research my proposal was to hire someone to listen to voicemails, type them and email them, as I couldn't see any way to do it, though in theory any good voi

Re: [asterisk-users] agi playback to execute say.conf settings

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ashik Ali Sent: Tuesday, September 14, 2010 2:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] agi playback to execute say.conf settings Hi

Re: [asterisk-users] Which 1.6 subversion is Stable one?

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nikhil Sent: Monday, September 13, 2010 10:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Which 1.6 subversion is Stabl

Re: [asterisk-users] Random File Name

Dan Journo wrote: > Hi, > > > > Im looking at using MixMonitor to record calls and I know that I need to > set the filename first. > > > > However, with the number of calls coming in, hard coding the filename > isnt an option. > > > > So I need to do something like this:- > > > >

Re: [asterisk-users] Speech To Text on linux with asterisk

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Tuesday, September 14, 2010 12:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Speech To Text on linux with asterisk Hi,

[asterisk-users] Random File Name

Hi, Im looking at using MixMonitor to record calls and I know that I need to set the filename first. However, with the number of calls coming in, hard coding the filename isnt an option. So I need to do something like this:- MixMonitor(RANDOMNUMBER.wav) But can't find a way to generate a ran

Re: [asterisk-users] SPA3102 FAX not working

Hi, I tried to send fax from Linksys to Grandstream by configuring openSER account.. that works fineonly when I send fax from Linksys to Asterisk I am not able to send On Thu, Sep 9, 2010 at 8:42 PM, Gopalakrishnan A.N wrote: > I am from India and I hope I have to use G711u...If I a

Re: [asterisk-users] Spontaneous reboots on asterisk 1.6.2.11

On 09/14/2010 02:30 PM, Jonas Kellens wrote: Hello list, has anyone else also noticed spontaneous reboots ?! I noticed this today and also yesterday. Can't really see if there is a fixed time between the reboots. Normally al of my SIP peers are registered. When I put up the CLI today I saw

[asterisk-users] Spontaneous reboots on asterisk 1.6.2.11

Hello list, has anyone else also noticed spontaneous reboots ?! I noticed this today and also yesterday. Can't really see if there is a fixed time between the reboots. Normally al of my SIP peers are registered. When I put up the CLI today I saw that a lot of SIP accounts where UNREACHABLE a

Re: [asterisk-users] Speech To Text on linux with asterisk

is it possible with lumenvox i will purchase liceance regards Dhaval On Tue, Sep 14, 2010 at 4:20 PM, Zeeshan Zakaria wrote: > In theory it should work but in real life it doesn't. Converting reliably > half an hour of speech into text is simply a dream. > > Zeeshan A Zakaria > > -- > www.ilove

Re: [asterisk-users] Speech To Text on linux with asterisk

In theory it should work but in real life it doesn't. Converting reliably half an hour of speech into text is simply a dream. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-14 4:52 AM, "Nickolay V. Shmyrev" wrote: В Втр, 14/09/2010 в 14:00 +0530, DHAVAL INDRODIYA пишет: > Thanks for upda

[asterisk-users] can asterisk accept "anonymous" register ?

Hi all, When i use asterisk-1.6, I must pre-write some extens to sip.conf or realtime db , then sip clients can register with asterisk. but now I want to do nothing with pre-writing extens to sip.conf or realtime db, when sip client register with asterisk, the asterisk accepte the register info an

Re: [asterisk-users] Speech To Text on linux with asterisk

В Втр, 14/09/2010 в 14:00 +0530, DHAVAL INDRODIYA пишет: > Thanks for update. > > is there any command for using sphinix to convert speech to text Yes, first of all make sure you compiled latest snapshot. Then run # sphinx_lm_sort < lm_giga_20k_nvp_3gram.arpa > lm_giga_20k_nvp_3gram.arpa.sorted

Re: [asterisk-users] Speech To Text on linux with asterisk

Thanks for update. is there any command for using sphinix to convert speech to text On Tue, Sep 14, 2010 at 1:18 PM, Nickolay V. Shmyrev wrote: > В Втр, 14/09/2010 в 01:55 -0400, Zeeshan Zakaria пишет: > > It is simply not possible, though it might be in the distant future. > > Let me respective

Re: [asterisk-users] Force ip disconnect after register?

"Bryant Zimmerman" writes: > Is there a way to force the connection to drop and reconnect after let's > say 50 attempts. Most firewalls have tools for removing specific connections from the connection table. Alternatively a switch to SIP/TCP might help, but I've never tried SIP/TCP with Asterisk

Re: [asterisk-users] Speech To Text on linux with asterisk

В Втр, 14/09/2010 в 01:55 -0400, Zeeshan Zakaria пишет: > It is simply not possible, though it might be in the distant future. Let me respectively disagree with you. It's perfectly possible even with open source tools. You can download pocketsphinx from http://cmusphinx.sourceforge.net To conver

Re: [asterisk-users] agi playback to execute say.conf settings

Hi danny, Shall we take it as agi bug ? Thanks, Ashik On Thu, Sep 9, 2010 at 7:10 PM, Danny Nicholas wrote: > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Ashik Ali > *Sent:* Thursday, S

[asterisk-users] 3xx redirect response list Noop and capture

Hi, I Just setup asterisk to send SIP calls to a SIP redirect server that response back with a list of destinations, if the first destination is not able to terminate the call, asterisk does not try the second , it just hang up. How can I Noop and capture the list inside the 3xx response?, for