Re: [asterisk-users] Moving from DSL to T1

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: Sunday, September 12, 2010 11:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Moving from DSL

[asterisk-users] Correct queue agi syntax in 1.6.2.11

Hello list, what is the correct syntax ? exten => s,n,Queue(${queuename}${timeout},cleanpickup.agi^${CHANNEL}) [Sep 13 10:23:58] WARNING[23551]: res_agi.c:886 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/cleanpickup.agi^SIP/329909007906-017a': File does not exist.

[asterisk-users] How to send SMS to Gigaset phones ?

Hi, Searching this list archives, I couldn't find a definitive answer to my question : how to send SMS to Gigaset phones ? My goal is to send Alert SMS such as "This phone system will be stopped in 5mn for maintenance" to every terminal (SIP phones and Gigaset DECT phones). (So at the moment, I'm

Re: [asterisk-users] Queue/Dial Recording - Capture answering channel name.

On 11 September 2010 20:36, Antonio Berrios wrote: > On 09/09/10 17:59, Steve Davies wrote: >> On 9 September 2010 17:52, Antonio Berrios >>  wrote: >> >>> Steve Davies wrote: >>> Hi, I am using 1.6.2.11, and I need to be able to include the name of the channel that answered a

[asterisk-users] High volume BLF - Suggestions?

Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed

[asterisk-users] Which voicemail file format is the most widely understood ?

Hi, In voicemail.conf you can choose among several file format (wav, wav49 and gsm) with which voicemail are saved. Which one the is the most widely read by Windows, Mac and Linux PC media players ? Suggestions ? Regards -- _ -

Re: [asterisk-users] Queue/Dial Recording - Capture answering channel name.

On 09/13/2010 10:11 AM, Steve Davies wrote: > On 11 September 2010 20:36, Antonio Berrios > wrote: >> On 09/09/10 17:59, Steve Davies wrote: >>> On 9 September 2010 17:52, Antonio Berrios >>> wrote: >>> Steve Davies wrote: > Hi, > > I am using 1.6.2.11, and I need to b

Re: [asterisk-users] Queue/Dial Recording - Capture answering channel name.

On 13 September 2010 11:07, Antonio Berrios wrote: > Gotcha. Yeah, I'm looking at implementing that (searching call > recordings by agent that took the call) here but since our asterisk call > recording is a separate server to the ones dealing with queues I'll be > looking at tie-ing the agent to

Re: [asterisk-users] Upgrade from 1.4 to 1.6 : problems with realtimemysql

This is a problem with extconfig.conf - not your res_ or cdr_ ones. In your case - extconfig.conf probably contained something like 'sippeers => mysql,MyDBase,sippeers'. The 'problem' is that the middle parameter is no longer for the database name - it is for the context in res_mysql.conf. So, t

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

We have good success with Grandstream, but even though we monitor 100 phones, I don't think we get the amount of NOTIFY messages you describe. But you should try them. Ron Op 13-09-10 11:56, Steve Davies schreef: > Hi, > > We have a user who is putting large call volumes through Asterisk, and > w

[asterisk-users] Changing voicemail.conf file format list

Hi, In voicemail.conf.sample, you can read this: format=wav49|gsm|wav ; WARNING: ; If you change the list of formats that you record voicemail in ; when you have mailboxes that contain messages, you _MUST_ absolutely ; manually go through those mailboxes and convert/delete/add the ; the message f

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

2010/9/13 Steve Davies > Hi, > > We have a user who is putting large call volumes through Asterisk, and > wants to BLF monitor up to 90 extensions. We are struggling to find a > handset that can keep up with Asterisk :) > > 1) Is there a handset that will do this? > 2) Is there a different (stand

Re: [asterisk-users] Which voicemail file format is the most widely understood ?

Hi, On 09/13/2010 10:57 AM, Olivier wrote: > Hi, > > In voicemail.conf you can choose among several file format (wav, wav49 > and gsm) with which voicemail are saved. > > Which one the is the most widely read by Windows, Mac and Linux PC media > players ? I use wav49. It is compatible with Window

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

On 13 September 2010 11:43, Olivier wrote: > > > 2010/9/13 Steve Davies [snip] >> Our test involves about 10 BLF-NOTIFY messages per second to each >> handset with a 5-second pause every 5 seconds. This will either crash >> or render unusable all of the following combinations: >> >> snom360 + 1 x

Re: [asterisk-users] Changing voicemail.conf file format list

Hi, On 09/13/2010 11:34 AM, Olivier wrote: > Hi, > > In voicemail.conf.sample, you can read this: > > format=wav49|gsm|wav > ; WARNING: > ; If you change the list of formats that you record voicemail in > ; when you have mailboxes that contain messages, you _MUST_ absolutely > ; manually go throug

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

Hello, Am 13.09.10 11:56, schrieb Steve Davies: > Hi, > > We have a user who is putting large call volumes through Asterisk, and > wants to BLF monitor up to 90 extensions. We are struggling to find a > handset that can keep up with Asterisk :) > > 1) Is there a handset that will do this? we onl

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

On 13 September 2010 12:16, Stefan Schmidt wrote: > Hello, > > Am 13.09.10 11:56, schrieb Steve Davies: >> Hi, >> >> We have a user who is putting large call volumes through Asterisk, and >> wants to BLF monitor up to 90 extensions. We are struggling to find a >> handset that can keep up with Aste

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

As a side note to this - do NOT try and use Aastra's - as they tend to crash after 50 BLF's! Also, could you please send me (perhaps off-list to a...@datavox.co.uk) your Yealink T28 findings - as I am a beta tester for them? Cheers Andy -Original Message- From: asterisk-users-boun...@li

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

On 13/09/10 11:03 PM, Steve Davies wrote: > On 13 September 2010 11:43, Olivier wrote: >> >> >> 2010/9/13 Steve Davies > [snip] >>> Our test involves about 10 BLF-NOTIFY messages per second to each >>> handset with a 5-second pause every 5 seconds. This will either crash >>> or render unusable all

Re: [asterisk-users] How to send SMS to Gigaset phones ?

On Mon, 13 Sep 2010, Olivier wrote: > Hi, > > Searching this list archives, I couldn't find a definitive answer to my > question : > how to send SMS to Gigaset phones ? > > My goal is to send Alert SMS such as "This phone system will be stopped in > 5mn for maintenance" to every terminal (SIP phon

Re: [asterisk-users] Moving from DSL to T1

On Mon, Sep 13, 2010 at 1:32 AM, Kevin Keane wrote: My numbers are from an AT&T DSL line in California, suburban San Diego > county, and just around the corner from the central office. So it is not the > distance (with DSL, the distance does make quite a difference). On the other > hand, there are

Re: [asterisk-users] How to send SMS to Gigaset phones ?

On Mon, Sep 13, 2010 at 3:29 PM, Gordon Henderson wrote: > On Mon, 13 Sep 2010, Olivier wrote: >> how to send SMS to Gigaset phones ? I dimly recall someone doing this and publishing a page of script... but where? Look using Google back from 2004-2008 something like "sms gigaset script" and I'll

Re: [asterisk-users] How to send SMS to Gigaset phones ?

2010/9/13 Gordon Henderson > > On Mon, 13 Sep 2010, Olivier wrote: > > > Hi, > > > > Searching this list archives, I couldn't find a definitive answer to my > > question : > > how to send SMS to Gigaset phones ? > > > > My goal is to send Alert SMS such as "This phone system will be stopped > in

Re: [asterisk-users] Problem with new AEX800 card dying because of interrupt problems

Christian Weeks writes: > Hello > I purchased an AEX800 card to replace the ageing cheap channel bank/T1 > card solution a few months ago, assuming that it would be a more robust > solution for my small scale phone system. However, it appears to be > anything but that. > > Originally implemented

[asterisk-users] Force ip disconnect after register?

Is there a way to drop a ip connection to asterisk after a number of register attempts. I have been having issues with hackers doing registration scanning against our server. We block their address at the fire wall but since asterisk does not force a drop of the connect after so many bad reg at

Re: [asterisk-users] Changing voicemail.conf file format list

On Monday 13 September 2010 06:07:09 Sebastian wrote: > On 09/13/2010 11:34 AM, Olivier wrote: > > In voicemail.conf.sample, you can read this: > > > > format=wav49|gsm|wav > > ; WARNING: > > ; If you change the list of formats that you record voicemail in > > ; when you have mailboxes that contain

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

Steve Grandstream has a new services GXP-21XX coming out they may work for your. We have been a beta tester and the BLF on these seem to work much better then the GXP-20XX units. I do not have the side cars in stock right now so I don't know how they work with it but you can put at least two fo

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

On Mon, Sep 13, 2010 at 11:22 AM, Bryant Zimmerman wrote: > Is there a way to drop a ip connection to asterisk after a number of > register attempts. > Not within Asterisk. Google fail2ban -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Fr

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

On Mon, Sep 13, 2010 at 11:22:33AM -0400, Bryant Zimmerman wrote: >Is there a way to drop a ip connection to asterisk after a number of >register attempts. Consider writing a filter for fail2ban [http://www.fail2ban.org/] that works on the Asterisk logs? -- _

Re: [asterisk-users] SIP softphones answer but do not connect...

On Mon, 2010-09-13 at 12:49 +1200, Matt Riddell wrote: > On 11/09/10 12:44 PM, Carlos Chavez wrote: > > The past few days I started having a problem with a small call center > > setup. All agents use Eyebeam 1.5 to receive calls from a queue. Eyebeam > > is > > configured to auto answer th

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

As I look to move our systems to version 1.8 I am looking at making a change from mySQL to PostgreSQL. I love mySQL but am getting very concerned about i'ts new owners. Should I be able to move all my realtime stuff to PostgreSQL is it fully supported with asterisk? Is there any down side to Po

Re: [asterisk-users] Correct queue agi syntax in 1.6.2.11

Hello, can anyone please tell me how I can give arguments to my AGI script ?! I think asterisk sees the name of the AGI + the channel as one filename, and of course this file then does not exist. Jonas. On 09/13/2010 10:26 AM, Jonas Kellens wrote: Hello list, what is the correct syntax ?

Re: [asterisk-users] SIP softphones answer but do not connect...

> > > > What strategy are you using for the Queue? > > > We are using Least Recent at the moment. Why would queue strategy > impact this? Carlos: I had similar issues, caused by a setting somewhere in the advanced section of eyeBeam. Something about "Disconnect if no audio received for x

Re: [asterisk-users] Queue/Dial Recording - Capture answering channel name.

On Mon, 2010-09-13 at 11:22 +0100, Steve Davies wrote: > On 13 September 2010 11:07, Antonio Berrios > wrote: > > Gotcha. Yeah, I'm looking at implementing that (searching call > > recordings by agent that took the call) here but since our asterisk call > > recording is a separate server to the on

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

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Monday, September 13, 2010 10:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] PostgreSQL is asterisk friendly with it? >

Re: [asterisk-users] SIP softphones answer but do not connect...

can you state your internet connection your agents are on?and one more thing.. how are the members positioned into the Queue? static? Dynamic? single station and call forwarding (find me follow me extension in the queue)? do you get call waiting override with Auto Answer? -- Tarek Sawah Integ

Re: [asterisk-users] Correct queue agi syntax in 1.6.2.11

On Mon, 2010-09-13 at 17:48 +0200, Jonas Kellens wrote: > Hello, > > can anyone please tell me how I can give arguments to my AGI script ?! > > I think asterisk sees the name of the AGI + the channel as one > filename, and of course this file then does not exist. > > In Asterisk 1.4 you

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

On Mon, 13 Sep 2010, Danny Nicholas wrote: > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant > Zimmerman > Sent: Monday, September 13, 2010 10:47 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asteri

Re: [asterisk-users] Queue/Dial Recording - Capture answering channel name.

On 13 September 2010 16:58, Carlos Chavez wrote: > On Mon, 2010-09-13 at 11:22 +0100, Steve Davies wrote: >> On 13 September 2010 11:07, Antonio Berrios >> wrote: >> > Gotcha. Yeah, I'm looking at implementing that (searching call >> > recordings by agent that took the call) here but since our as

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

On Mon, Sep 13, 2010 at 11:22:33AM -0400, Bryant Zimmerman wrote: > Is there a way to drop a ip connection to asterisk after a number of > register attempts. > > I have been having issues with hackers doing registration scanning against > our server. We block their address at the fire wall but s

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

On Mon, Sep 13, 2010 at 12:31:55PM -0400, Vince Vielhaber wrote: > > change from mySQL to PostgreSQL. > >> I love mySQL but am getting very concerned about i'ts new owners. > >> Should I be able to move all my realtime stuff to PostgreSQL is it fully [snippage and probably off topic] Why are you

Re: [asterisk-users] doing dnsmgr_lookup

At 11:38 PM 9/12/2010, you wrote: >my CLI is spammed with : > >[Sep 13 08:31:38]> doing dnsmgr_lookup for 'ssw6.itsp.tld' > > >How can I turn this off ?! I just change the 4 in the source code to 14 so it doesn't show up till you add a lot of Vs. It only occurs in one place in the code so

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

I agree with Steve, I've had a couple of people speak with me about concerns of MySQL's new ownership, but I stand by the same reasons Steve doesMySQL will not be going anywhere, and if it does, someone'll just fork and keep it going under a new name. On Mon, Sep 13, 2010 at 11:46 AM, Steve

Re: [asterisk-users] How to send SMS to Gigaset phones ?

Hi Olivier, I remember having had a similar discussion a few years ago. I will paste my postings from around May 2007 further down. First, I did not try sending SMS over VOIP to the phone, just over Voip to an ATA and then over analogue line (or ISDN) to the phone. So I have no idea wether the ne

Re: [asterisk-users] SIP softphones answer but do not connect...

On Mon, 2010-09-13 at 15:59 +, Tarek Sawah wrote: > can you state your internet connection your agents are on? > and one more thing.. how are the members positioned into the Queue? > static? Dynamic? single station and call forwarding (find me follow me > extension in the queue)? do you get cal

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

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 -- _ -- Bandwidth and C

Re: [asterisk-users] Correct queue agi syntax in 1.6.2.11

On 09/13/2010 06:01 PM, Carlos Chavez wrote: On Mon, 2010-09-13 at 17:48 +0200, Jonas Kellens wrote: Hello, can anyone please tell me how I can give arguments to my AGI script ?! I think asterisk sees the name of the AGI + the channel as one filename, and of course this file then does not

Re: [asterisk-users] A way to check against a list of numbers?

Hose writes: > The most straightforward way would be to just define explicit patterns. > Obviously that works, but doesn't seem scalable in terms of maintenance. I don't think that maintaining the list in the dial plan is all that bad, actually. Dump it in its own context and file... If that is

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

"Bryant Zimmerman" writes: > As I look to move our systems to version 1.8 I am looking at making a > change from mySQL to PostgreSQL. > > I love mySQL but am getting very concerned about i'ts new owners. > Should I be able to move all my realtime stuff to PostgreSQL is it fully > supported with a

Re: [asterisk-users] Correct queue agi syntax in 1.6.2.11

On Mon, Sep 13, 2010 at 08:15:34PM +0200, Jonas Kellens wrote: > [Sep 13 20:14:59] -- Launched AGI Script > /var/lib/asterisk/agi-bin/cleanpickup.agi > [Sep 13 20:14:59] opruimenpickup.agi: Failed to execute > '/var/lib/asterisk/agi-bin/cleanpickup.agi': Permission denied So check that /

Re: [asterisk-users] Correct queue agi syntax in 1.6.2.11

No that is not the problem, nor was it the question. I found the solution. Apparently you need to place the AGI-script and its arguments between " ", but the arguments still need to be separated by a comma. Example : exten => s,n,Queue(queuename,"myscript.agi,arg1,arg2") If anyone wan

Re: [asterisk-users] doing dnsmgr_lookup

Hello, anyone on this list knows how to turn these messages off please ?! I have in sip.conf : srvlookup=no and in dnsmgr.conf : [general] enable=no; enable creation of managed DNS lookups ; default is 'no' ;refreshinterval=1200; refresh managed DNS lookups every se

Re: [asterisk-users] A way to check against a list of numbers?

i have this scenario where i have a marketing department calling USA numbers excessively and sometimes the leads contain duplicate numbers OR "duplicate customers with different numbers" on the other hand we have some numbers that are black listed the destination should be checked and caller s

Re: [asterisk-users] Moving from DSL to T1

On Mon, 2010-09-13 at 00:32 -0700, Kevin Keane wrote: > > > Latency also is the reason VoIP does not work at all over satellite > > connections even though they tend to have plenty of bandwidth. > Please define "does not work at all over satellite" ??? > Sure, it is not studio HIFI quality, but

Re: [asterisk-users] doing dnsmgr_lookup

On Mon, Sep 13, 2010 at 2:53 PM, Jonas Kellens wrote: > anyone on this list knows how to turn these messages off please ?! > *CLI> core set verbose 0 -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- ___

Re: [asterisk-users] Moving from DSL to T1

On Mon, Sep 13, 2010 at 1:12 PM, Hans Witvliet wrote: > No these are also geo-stationary (same altitude, so same delay), > commercial and military satelites, > Yes, exactly. Geostationary satellites have been used for telephone for ages (and are still used for remote areas - they have advantag

Re: [asterisk-users] Asterisk 1.6 and fax

Hello! I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5. When i try to receive fax I get: [Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel 'SIP/crocus-ua-0004' refused to negotiate T.38 [Sep 13 00:46:02] WARNING[3283]: app_fax.c:223 phase_e_handler

Re: [asterisk-users] Asterisk 1.6 and fax

On Mon, Sep 13, 2010 at 4:33 PM, Stanislav Korsei wrote: > Hello! > I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5. > When i try to receive fax I get: > [Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel > 'SIP/crocus-ua-0004' refused to negotiate T

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

On 09/13/2010 10:22 AM, Bryant Zimmerman wrote: > Is there a way to drop a ip connection to asterisk after a number of > register attempts. > > I have been having issues with hackers doing registration scanning > against our server. We block their address at the fire wall but since > asterisk does

Re: [asterisk-users] Moving from DSL to T1

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: Monday, September 13, 2010 12:13 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Moving from DSL to T1 On Mon, 2010-09-

Re: [asterisk-users] Asterisk 1.6 and fax

On 09/14/2010 04:33 AM, Stanislav Korsei wrote: > Hello! > > I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5. > When i try to receive fax I get: Why install 0.0.5? Its ancient. the world has moved on. > > [Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: chan

Re: [asterisk-users] Moving from DSL to T1

On 09/14/2010 04:23 AM, Joel Maslak wrote: > On Mon, Sep 13, 2010 at 1:12 PM, Hans Witvliet > wrote: > > No these are also geo-stationary (same altitude, so same delay), > commercial and military satelites, > > > Yes, exactly. Geostationary satellites have been

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

Hi all, I would like to install asterisk as my home pbx, Anyone can suggest which sub version of 1.6 is stable? Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? J

Re: [asterisk-users] Polycom dhcp boot

Your string and boot-option look good. In the SonicWall config, its a two step process: - create the new boot option under "DHCP Server" menu > "Advanced" button > "Add Option" - assign it to your lease scope under "DHCP Server" menu > "DHCP Server Lease Scopes" section > "Edit" button > "Advanc

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

Hi, Is it possible to record say 30 seconds of audio and then have LumenVox convert to text ? or any available tool open source for speech to text . Regards Dhaval -- _ -- Bandwidth and Colocation Provided by http://www.api-

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

On Tue, Sep 14, 2010 at 1:01 AM, DHAVAL INDRODIYA wrote: > Is  it  possible to record say 30 seconds of audio and then have LumenVox > convert to text ? > ASR, yes. http://www.digium.com/en/products/software/lumenvox.php -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@poly

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

Thanks Paul, i think still i have some problem to understand , i mean to say that i have 30 minutes audio file in WAV format and i wnat its text here are the scenario . -> Call comes in -> start recording -> call remains for 30 minutes -> stop recording -> convert wav file audio to text. is this

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

It is simply not possible, though it might be in the distant future. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-09-14 1:50 AM, "DHAVAL INDRODIYA" wrote: Thanks Paul, i think still i have some problem to understand , i mean to say that i have 30 minutes audio file in WAV format and i wna

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

Hi, With my Gigaset C470IP (with latest 02223 firmware), I can't find a way to access SMS settings from web configuration app or using a handset. Has someone been more successful without using auto-configuration mode ? (For instance, manual says an SMS entry is showing on handset screen but as I