Re: [asterisk-users] bash: asterisk: command not found

On 07/12/16 04:56, christopher kamutumwa wrote: Ive installed asterisk 14.2 on centos 6.8 but i am not able to start it below is what am executing and those are the errors anything am doing wrong? It doesn't look like it is installed to me... Check the install actually worked etc. I've never

Re: [asterisk-users] Just got defrauded - how do I block calls which contain a dash (RegEx noob question)

On 28/10/16 16:38, Markus wrote: I'm using Asterisk2Billing (v2.0.16) and it appears to have an annoying bug. When there are rates for e.g. 44 (UK landline) and 44870 (UK premium) and a fraudster manages to somehow dial 44-870 instead of 44870 the rate for 44 will match, not the one for 44870.

Re: [asterisk-users] Removing mailbox and password prompt for voicemail

On 01/08/16 09:08, Nabeel wrote: I am yet to test this behaviour in Asterisk during the Unavailable/Busy message. However, if this is the case, then this seems to be an illogical security hole in Asterisk's design. Why does Asterisk allow accessing another person's mailbox by pressing the '*'

Re: [asterisk-users] implementing asterisk call center.

On 06/04/16 20:58, Goke Aruna wrote: Can someone help me with a kind of howto build call center around asterisk with all the necessary features like CTI, call recordings, call spying, real time monitoring etc? What is your budget? I'm sure there are many contractors who can help. Steve --

Re: [asterisk-users] SIP trunk with whatsapp

On 28/03/16 12:46, bilal ghayyad wrote: Does anyone has information if possible to setup SIP trunk with whatsapp? How can we let asterisk send and receive calls from whatsapp? I don't think you can. Whatsapp is a closed system. Steve --

Re: [asterisk-users] Voice recognition IVR Is it possible?

On 22/02/16 23:58, Frank wrote: On Tue, 2016-02-23 at 00:43 +0100, Laszlo wrote: ... Speech API key from Google Yes... OK... but... where and how can I obtain this API Key? Google?... Steve -- _ -- Bandwidth and

Re: [asterisk-users] Authenticate() 11.21.0

On 10/02/16 14:20, Jerry Geis wrote: I am trying to use Authenticate() in the dialplan for something other than "my password". The message says "Please enter YOUR password followed by the pound key". I'm not using this for my password. Is there any way to change the message to "please enter the

Re: [asterisk-users] How to deal with error messages passed as Early Media

On 03/02/16 15:29, Olivier wrote: 2016-02-03 15:59 GMT+01:00 Steve Howes <steve-li...@geekinter.net <mailto:steve-li...@geekinter.net>>: On 03/02/16 14:41, Olivier wrote: How can I best deal with error messages passed as Early Media. Tell the ITSP to giv

Re: [asterisk-users] How to deal with error messages passed as Early Media

On 03/02/16 14:41, Olivier wrote: How can I best deal with error messages passed as Early Media. Tell the ITSP to give you proper signaling, if they wont then get a new ITSP. I suspect if they can't handle this correctly, there will be a lot more they're doing wrong as well. Long term you'll

Re: [asterisk-users] Fwd: Sublime Text License Key

Wonder what happens when an entire mailing list tries to use that key?... On 05/10/15 15:28, Optical Phoenix wrote: -- Forwarded message -- From: *Sublime HQ Pty Ltd* > Date: Wednesday, July 25, 2012 Subject: Sublime Text

Re: [asterisk-users] Fwd: Sublime Text License Key

On 05/10/15 16:18, Mitul Limbani wrote: The company making sublime text gets few thousands of dollars of notional loss :) I was thinking more about if they'd built in software activation type stuff. But yea, stealing bad etc too. Steve --

Re: [asterisk-users] uptime

On 15 Feb 2011, at 03:39, Jeff LaCoursiere wrote: minipbx*CLI show uptime System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds Last reload: 8 hours, 3 minutes, 51 seconds What's the highest current 'genuine' one on-list?.. klein*CLI core show uptime System uptime: 2

Re: [asterisk-users] Hide the plain text password

On 15 Feb 2011, at 13:17, Richard Kenner wrote: Of course not! It would be useless if that were the case: the whole point here would be that you need the master encryption key. Here's a possible design: - There's optionally a file in the config directory called master_key. It contains

Re: [asterisk-users] On-Hold Music

On 11 Feb 2011, at 22:37, Danny Nicholas wrote: In 500 words or less (if possible), please explain what is a legal music-on-hold file? Depends on the country, and what licence you posses. Googling 'countryname hold music regulations' may help. S--

Re: [asterisk-users] ${HANGUPCAUSE} in CDR

Steve Howes SMTP to Google proxy Inc -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] forward calls by the ports

On 8 Feb 2011, at 14:52, mehran khajavi wrote: i searched a lot but i couldn't find the answer . i have two openvox(fxo/fxs) card so I have 24 ports! Ok! on first card i have 12 fxs and on the second i have 12 fxo i want to then one person calling from dahdi/13 forward it to dahdi/1 when

Re: [asterisk-users] faxter

On 30 Jan 2011, at 09:21, Pezhman Lali wrote: Faxter is an opensource email to fax gateway, please check it, let me know if any bug. Only bug i can see is the attitude of the developer... As for the bugs, having the config variables liberally scattered throughout the script makes it's use

Re: [asterisk-users] Unable to insert cdr-data into mysql-DB

On 25 Jan 2011, at 09:36, Andrew Thomas wrote: Try changing 'hostname=127.0.0.1' to 'hostname=localhost' in the cdr_mysql.conf. I seem to remember a problem I had when '127.0.0.1' and 'localhost' didn't marry up never did find out why. I believe localhost means it can use a socket, where as

Re: [asterisk-users] Asterisk stops responding

On 22 Jan 2011, at 18:02, Carlos Chavez wrote: Cannot allocate memory Have you tried looking at memory? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] Mailing list question 2

On 20 Jan 2011, at 17:13, Andrew Thomas wrote: Sorry about this - testing this disclaimer problem :) I can give you a POP3 account on my server if it stops you spamming the list?.. S -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Continuously core dumping of 1.8 on SLES

On 17 Jan 2011, at 11:29, Hans Witvliet wrote: Missing something obviously, core dump / backtrace? ;) Might be worth knocking a few of the modules out that were listing errors to see if any of them are causing it. It's possible something not loading isn't being handled gracefully. S --

Re: [asterisk-users] How to check a number online or offline

On 10 Jan 2011, at 10:17, Phuong Hoang wrote: Thanks enkillar, but this is`nt thing that i need. I want to check number online, offline or unreachable on asterisk using AMI(Asterisk Manager Interface) by java but i have`nt found a solution yet. I hope you can help me do this. Thanks in

Re: [asterisk-users] How to check a number online or offline

On 10 Jan 2011, at 10:37, Phuong Hoang wrote: I found the link you have just sent to me but it do`nt help me to resolve this. Can you say clearlier for me? Not really. It's a list of manager commands. There is 'SIPshowpeer' which will work for sip stuff. Try the command 'Command' action and

Re: [asterisk-users] Call sip:u...@domain.com?

On 21 Dec 2010, at 14:20, A J Stiles wrote: Well, every Free and Open Source telephony system is using Asterisk (and Linux) under the bonnet. The differences are in the user configuration tools. Uh, no? S -- _ --

Re: [asterisk-users] Mail Integration

On 13 Dec 2010, at 14:25, Danny Nicholas wrote: (god forbid) postal mail Haha, I'm kind of tempted to write an app_cups module to print envelopes ;) S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] Push central phone book to phones

On 7 Dec 2010, at 11:35, Jonas Kellens wrote: When on a public server, I find this insecure. Then secure it? Tie down by IP address, or some phones support the username:password@ in a URL. S -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Abandon events in cdr

On 3 Dec 2010, at 13:47, Rodrigo Lang wrote: unansweredy = yes Remove the extra y. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

Re: [asterisk-users] Asteris 1.8 and mISDN - 'mISDN' (cause 66 - Channel not implemented)

On 1 Dec 2010, at 10:18, Michael Nausch wrote: If I start asterisk 1.8 with service asterisk start or /etc/init.d/asterisk start, I can't load chan_misdn.so If I run asterisk 1.8 as root via asterisk -vvvc I can access my ISDN-card and I be able to dial out to my PSTN provider! ;) File

Re: [asterisk-users] TCP port, VPN and resolving the cutting voice problem

On 30 Nov 2010, at 09:28, bilal ghayyad wrote: If I ran IAX in TCP port, and in case my network was having a lot of users doing browse on the internet and downloading, so in that case and if the IAX used TCP port, so the voice will be better than using UDP (because in TCP the lost packets

Re: [asterisk-users] Asteris 1.8 and mISDN - 'mISDN' (cause 66 - Channel not implemented)

On 30 Nov 2010, at 09:47, Michael Nausch wrote: I tried to configure Asterisk 1.8 on one of my test-hosts. I've installed from centos-asterisk.repo (http://packages.asterisk.org/centos/$releasever/tested/$basearch/): snip [Nov 30 10:35:53] WARNING[7281]: channel.c:5353 ast_request: No

Re: [asterisk-users] Stability..

On 28 Nov 2010, at 22:26, dotnetdub wrote: It could be an extension name Where is the error trapping if this is the case.. Who writes this shit? A dedicated bunch of volunteers who don't appreciate you being a dick about bugs, which you report without so much as a log entry or a core

Re: [asterisk-users] How to register SIP phone on Asterisk 1.6.2.14 on Centos 5.5 64bit

On 18 Nov 2010, at 10:36, Phuong Hoang wrote: I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but not successful, Can anyone help me to do it? How is this different to the other two posts? Please stop repeatedly sending messages! If nobody replies you're probably not

Re: [asterisk-users] Fwd: Register SIP on Asterisk 1.6.2.14 on Centos 5.5 64bit

On 18 Nov 2010, at 10:33, Phuong Hoang wrote: I registered sip phone(X-Lite) on Asterisk 1.6.2.14 on CentOS 5.5 64 bit but not successful, Can anyone help me to do it? Given that you haven't given any error messages, any logs, or your sip.conf, or the manner in which it is not working

Re: [asterisk-users] install

On 7 Nov 2010, at 20:59, Thomas Perron wrote: I have installed Asterisk before w/ no issues but while trying today (1.6.2.13 and centors 5.4) I receive the following at the CLI: The configure script must be executed before running 'make'. Please run ./configure. Any

Re: [asterisk-users] MixMonitor

On 5 Nov 2010, at 01:22, Mickael MONSIEUR wrote: Have you noticed a marked increase in CPU load when using MixMonitor? Since when? 1.6.2.9-1? 1.6.2.8? 1.0? S -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] What is digium doing on port 113?

On 31 Oct 2010, at 01:29, Joel Maslak wrote: Probably doing an ident lookup when you send mail to the list. Standard sendmail behavior. Agreed. Nothing to worry about. S -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Auto provisioning from public server

On 26 Oct 2010, at 16:31, Jonas Kellens wrote: has anyone experience with auto provisioning IP-phones on different locations through a central public provisioning server ? You use http or https ? What handset? That's rather what controls your options. Some support HTTPS with client

Re: [asterisk-users] Dial Plan Conf

On 21 Oct 2010, at 10:16, Jigar Joshi wrote: I have attached the dial plan file. In what format? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] Needed in Dallas, Texas, Network Voice Engineer and Field Service Install Technician

On 21 Oct 2010, at 15:56, JR Richardson wrote: These are full time positions in Dallas, no telecommuters please. A very vast majority of people on here are not in Dallas (and indeed probably a majority in the US). So stop filling their mailboxes with this crap. Incase you hadn't noticed

[asterisk-users] SIP Blacklisting

Hi, Given the recent increase in SIP brute force attacks, I've had a little idea. The standard scripts that block after X attempts work well to prevent you actually being compromised, but once you've been 'found' then the attempts seem to keep coming for quite some time. Older versions of

Re: [asterisk-users] SIP Blacklisting

On 21 Oct 2010, at 16:54, Jeff LaCoursiere wrote: I'll subscribe, that is for sure. What is the best way to dist the blacklist? iptables include file? Or something more integrated to asterisk... just thinking off the top of my head that a module that vetted inbound connections against

Re: [asterisk-users] SIP Blacklisting

On 21 Oct 2010, at 17:03, Zeeshan Zakaria wrote: But the problem is how to make sure that only legitimate users are contributing to this list. Contributors to this list somehow need to verify to an admin that they are not hackers, and this the hard part. I was thinking of having a threshold

Re: [asterisk-users] SIP Blacklisting

On 21 Oct 2010, at 17:32, Jeff LaCoursiere wrote: I agree in principle - some cron job pulling the list by http would certainly be simple. But just to continue my thoughts to the brick wall, I don't see a lookup adding latency to the call other than what should be a very brief addition to

Re: [asterisk-users] asterisk router

On 7 Oct 2010, at 23:57, steve casto wrote: A Crisco RVS4000 installed now has real problems with Sip, one-way audio and throughput not up to the WAN speed. ALG? (Assuming you mean Cisco..) -- _ -- Bandwidth and Colocation

Re: [asterisk-users] MYSQL ADDON INSTALLATION ERROR

On 6 Oct 2010, at 11:35, Rizwan Hisham wrote: Hi All, Please refresh my memory. I am trying to install asterisk after 2 years. I hav'nt used it since 2008 (version 1.4.2). Now I am trying to install 1.8.0-rc2 on centos 5.5 but getting the following errors. snip Plz help. You need

Re: [asterisk-users] Implementing more than one asterisk instance in the same hardware machine?

On 5 Oct 2010, at 15:13, Gordon Henderson wrote: $ /home/asterisk1/usr/sbin/asterisk -g for first asterisk $ /home/asterisk2/usr/sbin/asterisk -g for second asterisk I did that before I moved to LXC, but you can't use the standard port 5060 for all instances, only one - might be OK in

Re: [asterisk-users] Asterisk/Realtime and MySQL

On 1 Oct 2010, at 09:52, Phibee Network Operation Center wrote: That's work and in my extension.conf, i have: [as5300-incoming] switch = Realtime and in extconfig.conf extensions = mysql,general,VOIP_Extensions A lot of Extension are into the table VOIP_Extensions.

Re: [asterisk-users] should trixbox system hang when ISP dropsconnection?

On 24 Sep 2010, at 16:09, Danny Nicholas wrote: The BOBW solution I would suggest is that you run your Trixbox/Asterisk using a local DCHP provider/server so you aren't as vulnerable to how efficient your ISP is at staying up. DNS. Not DHCP. S --

Re: [asterisk-users] Digest Username/auth name mismatch ‏

On 23 Sep 2010, at 17:29, t. k wrote: Isn't there any way to configure the username in the hardphone to be just ? Yes.there is no way to cofigure as in the hardphone.It will cost and spend time a lot to implement Then I think the short answer is that it's not compatible. Steve --

Re: [asterisk-users] http://www.asterisk.org/downloads naming schema

On 22 Sep 2010, at 16:45, Klaus Darilion wrote: Since some time the download of the newest Asterisk does not contains the version number anymore, but is just called asterisk-1.4-current.tar.gz This gives me a tarball where I do not know the version without looking into the tarball.

Re: [asterisk-users] Realtime semi-colon

On 16 Sep 2010, at 12:56, Andrew Thomas wrote: Does anyone know how to send * a semi-colon from a realtime database. I know that * uses the semi-colon as a 'seperator' - but I need to be able to use one in a command. I know I can use \; in the non-realtime configs, but this doesn't work in

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

On 15 Sep 2010, at 13:22, Jonas Kellens wrote: I have indeed found the core file in /tmp (that is where 'locate' does not look huh...) 'updatedb'? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

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] 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

Re: [asterisk-users] Wanted: UK-specific hardware recommendations (FXO and FXS)

On 3 Sep 2010, at 10:07, Roger Burton West wrote: Also: I've heard good things about the PAP2T for getting analogue handsets to talk to a VoIP server. But all the ones I can see on eBay are PAP2T-NA models. Will these work with British handsets? (Obviously with a plug adaptor to put the BT

Re: [asterisk-users] Asterisk routing to SoftSwitch

On 1 Sep 2010, at 10:30, Pratik Shrestha wrote: Any Idea?? Read http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf But I'm guessing you knew that and are just after getting someone else to do the work... Just create a catch-all pattern to match anything

Re: [asterisk-users] Yes it is a dimensioning question! Atom CPU

On 31 Aug 2010, at 16:37, jmilli...@sentinelcommunications.com wrote: I am looking at an Atom D510 (dual core 1.6GHz, 1M cache) to run maybe 25 to 30 extensions, 4 or 5 calls at once(maybe as high as 7 simultaneous calls), g729 all the way through Sounds fine to me. Reckon you could do that

Re: [asterisk-users] Yes it is a dimensioning question! Atom CPU

On 31 Aug 2010, at 17:58, jmilli...@sentinelcommunications.com wrote: On 31 Aug 2010, at 16:37, jmilli...@sentinelcommunications.com wrote: I am looking at an Atom D510 (dual core 1.6GHz, 1M cache) to run maybe 25 to 30 extensions, 4 or 5 calls at once(maybe as high as 7 simultaneous calls),

Re: [asterisk-users] Yes it is a dimensioning question! Atom CPU

On 31 Aug 2010, at 18:10, Andrew Latham wrote: Sounds fine to me. Reckon you could do that on a toaster ;) Thanks, I needed to clean this keyboard anyway Hehe. It's true though. I was amazed what our atom boards would do. We even chucked transcoding/conferencing at them and they worked

Re: [asterisk-users] Codec choice

On 19 Aug 2010, at 09:14, Deepika Nijhawan wrote: Does anyone has an idea how to tell asterisk to use codec A for first 50 calls and then codec B for rest of the calls. You could create two separate trunks, one for each codec? S --

Re: [asterisk-users] install asterisk

On 13 Aug 2010, at 14:08, Albert Bonomo wrote: How come nowhere in the internet, nor in Digium.com docs, blogs, or whatever, anybody mention that yum install is available ? Why nobody ever make a small note telling that asterisk is available from repositories to install, and that is so easy

Re: [asterisk-users] Asterisk 1.8.0-beta1 Connectedline

On 24 Jul 2010, at 13:21, unsero...@aol.com wrote: Hi, i just tried to use the CONNECTEDLINE() feature but it does not work, at least with my softphones (zoiper, 3CX, Xlite) in sip.conf under general I have: trustrpid = yes sendrpid = rpid,pai rpid_update = yes in extensions.conf

Re: [asterisk-users] BLF - Realtime Asterisk

On 16 Jul 2010, at 09:17, Danny Dias wrote: [Jul 15 18:43:14] WARNING[15867]: chan_sip.c:15766 handle_request_subscribe: SUBSCRIBE failure: unrecognized format: 'multipart/related' pvt: subscribed: 0, stateid: -1, laststate: 0, dialogver: 0, subscribecont: 'pbx9', subscribeuri: '' Looks like

Re: [asterisk-users] MyFuel Express FO - Shortcomings

Did you mean to send this to a mailing list?.. S On 13 Jul 2010, at 13:33, Alphonse Ogulla wrote: Re-sent copying UNON and Expand Technologies. Apologies for the omission. Rgds, Alphonse On Tue, Jul 13, 2010 at 3:27 PM, Alphonse Ogulla aogu...@gmail.com wrote: Dear Esther, The

Re: [asterisk-users] Remote-Party-ID party=called

On 12 Jul 2010, at 14:09, Jonas Kellens wrote: I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. Am I missing something or is this waht CALLERID(name) and sendrpid is for?.. S --

Re: [asterisk-users] Remote-Party-ID party=called

On 12 Jul 2010, at 16:35, Jonas Kellens wrote: On 07/12/2010 05:01 PM, Steve Howes wrote: On 12 Jul 2010, at 14:09, Jonas Kellens wrote: I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. Am I missing something

Re: [asterisk-users] ARA : Realtime or not ?

On 6 Jul 2010, at 10:34, Jonas Kellens wrote: what is the use of realtime SIP peers when you always need to reload the sip configuration as if you were just putting your SIP peers in sip.conf ?? Did you enable caching by any chance? S --

Re: [asterisk-users] Remote Party ID issue

On 2 Jul 2010, at 12:29, John Novack wrote: regardless, people will post either way, and wasting archive space complaining about either one is pointless. I was mainly pissed off about him directly replying to people (i.e. me) rather than the list. troll It was you lot that started the

Re: [asterisk-users] Remote Party ID issue

On 1 Jul 2010, at 15:52, unsero...@aol.com wrote: [Jul 1 16:30:50] ERROR[3954]: pbx.c:2902 ast_func_write: Function CONNECTEDLINE not registered Same happens trying function CALLEDID. I am using Asterisk 1.6.1.20. What do i have to do to use this function or alternatively the

Re: [asterisk-users] Remote Party ID issue

On 1 Jul 2010, at 16:25, unsero...@aol.com wrote: Sorry, what does this mean? Only in trunk? If you look in the post you quoted This feature is in Asterisk trunk and will be present in the upcoming 1.8 release. First sentence. S --

Re: [asterisk-users] Remote Party ID issue

On 1 Jul 2010, at 16:56, unsero...@aol.com wrote: Sorry, i wanted to know what is in trunk means. So it seems to mean is in the pipeline for the next version. DON'T reply to people off list. And stop bloody top posting. Steve --

Re: [asterisk-users] Echo problem in VoIP-calls

On 30 Jun 2010, at 13:48, Gareth Blades wrote: By ITSP do you mean a SIP provider? ITSP: Internet Telephony Service Provider S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

Re: [asterisk-users] sip add header

On 28 Jun 2010, at 13:08, Jerry Geis wrote: It works fine with I call the SIP phone directly - however - when I first call the Local channel - then Dial the SIP phone the SIPADDHEADER doesnt seem to do anything. Are you adding the header before or after you dial the local channel? S --

Re: [asterisk-users] Pickup a ringing Queue member

On 28 Jun 2010, at 15:36, Jonas Kellens wrote: Does this mean I have a patched asterisk ? (I ask this because some applications require a non-patched asterisk version) Yes. What is then the unpatched version of Asterisk 1.4.30 ?? The one you have before you apply the patch?.. --

Re: [asterisk-users] Very strange registration problem

On 24 Jun 2010, at 12:49, Jonas Kellens wrote: It seems as if some SIPaccounts could register and others could not. I don't think a firewall distinguishes between phone brands or SIP accounts. Alas 'stabbing in the dark' is all we can do until you actually provide some information for us.

Re: [asterisk-users] one for your filters

On 23 Jun 2010, at 18:39, Steve Edwards wrote: Ouch. 82.0.0.0/8 is on my block list, available at: http://www.sedwards.com/class-a-block-list Would advise people in the UK do not use that list... 82.0.0.0/8 would block a reasonable chunk of my users for starters.. Steve --

Re: [asterisk-users] one for your filters

On 23 Jun 2010, at 19:26, Steve Howes wrote: On 23 Jun 2010, at 18:39, Steve Edwards wrote: Ouch. 82.0.0.0/8 is on my block list, available at: http://www.sedwards.com/class-a-block-list Would advise people in the UK do not use that list... 82.0.0.0/8 would block a reasonable

Re: [asterisk-users] Slightly OT: Cisco SPA525G and network errors

On 17 Jun 2010, at 15:58, Mike wrote: I have a Cisco SPA525G latest firmware, and very often when I attempt a transfer I get a network error message when I press Dial on the transfer. I never get that erroron a simple call out Asterisk is configured for that phone exactly the same as

Re: [asterisk-users] Out of Office

Orlando, FL 32819 m...@accessgate.net Office Toll Free: (888) 227-9337 Fax: (407) 352-2717 From: Steve Howes steve-li...@geekinter.net Sent: Thursday, June 10, 2010 4:26 AM To: d...@accessgate.net, m...@accessgate.net, jsny...@accessgate.ne Subject: Re: [asterisk-users] Out of Office

Re: [asterisk-users] reloading realtime sip peers

On 8 Jun 2010, at 16:40, Jonas Kellens wrote: I noticed that changes to realtime sip peers are not applied until a 'reload'. A 'sip reload' does not make any changes to realtime sip peers. sip prune ? S -- _ -- Bandwidth and

Re: [asterisk-users] how to get call duration

On 3 Jun 2010, at 14:24, Necati Demir wrote: I want to ask how to get call duration. Go on then When you do ask the question you might want to include a few details. Are you trying to get call duration during a call? If so then the cli will help 'core show channels'. If it's after the

Re: [asterisk-users] Voicemail : mail attachment to multiple mail-addresses

On 1 Jun 2010, at 16:53, Jonas Kellens wrote: Sounds... p Perhaps you could contribute a patch? ;) S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] Softphones on thin clients...

On 20 May 2010, at 18:35, Carlos Chavez wrote: I am worried about conflicts when running 10 softphones on the same server since they will all try to use por 5060. And the fact most terminal services servers/clients still don't support audio input.. only output.. S --

Re: [asterisk-users] OT: NAT in SPA922

On 6 May 2010, at 14:16, Sebastian Milioto wrote: Ok..So what ip phone model do NAT? I think you'd struggle to find one. If it's a requirement you're probably doing something wrong... S -- _ -- Bandwidth and Colocation

Re: [asterisk-users] OT: NAT in SPA922

On 5 May 2010, at 14:39, Sebastian Milioto wrote: However, when I connect a PC to that port, SPA922 works as bridge. Anybody can confirm SPA922 can NAT a PC connected to its LAN port? Does exist such LAN tab for setting up parameters as port forwarding? (by the way, version is 5.1.15(a).

Re: [asterisk-users] Channel failover

On 4 May 2010, at 03:44, Jack Bates wrote: We recently got VoIP, so when we make a call, Asterisk should first try to make the call with VoIP, but in case either our VoIP or our internet service are down, Asterisk should then try to make the call with our old school analog phone line Well,

Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286

On 30 Apr 2010, at 09:41, Vieri wrote: As far as having an internal fan for cooling, I don't know if that's actually better... In general, these devices shouldn't need to rely on mechanical cooling which tends to fail in time (sure, you can open the case and replace it but that's extra

Re: [asterisk-users] Issue with (pattern) matching extension

On 29 Apr 2010, at 22:56, Leif Madsen wrote: Danny Nicholas wrote: Good snippet, Leif. It's easier to read 100 threads on this forum than the 100 pages of the infamous Asterisk Book PDF. Infamous? Ouch :) He's insulting our holy book! Stone him! ;) S --

Re: [asterisk-users] Dial plan question.

On 28 Apr 2010, at 06:53, Aditya Kumar wrote: exten = bob,1,Dial(SIP/${exte...@ext-sip,20) Where did you define EXTERN? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] Asterisk choking on voice messages announcements

On 22 Apr 2010, at 00:36, bruce bruce wrote: Opened pseudo dahdi interface, measuring accuracy... 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434% -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770% 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622%

Re: [asterisk-users] Help with FastAGI server in Windows

On 19 Apr 2010, at 17:00, Edwin Quijada wrote: Hi! I am trying to do a FastAGI server in windows. I am using the example from their page but I dont get anything. Anybody here has experienced with Fastagi in windows and perl that give a rigth direction to do this. I have experience with AGI

Re: [asterisk-users] Realtime changes not reflected realtime

On 17 Apr 2010, at 10:25, Jonas Kellens wrote: When changing the secret, the old secret is still the one to use until a sip reload. When changing the name, the old name is still the one to use for registrations until a sip reload. So it's being cached? Does 'sip prune realtime all' clear

Re: [asterisk-users] Delay the HungUp

On 16 Apr 2010, at 14:39, cbulist wrote: We need to delay the HungUp because some calls that we dial are so short (3 or 4 seconds) and our provider requests 8 seconds. That is the reason Sounds like you need a decent provider ;) S --

Re: [asterisk-users] Is svn.asterisk.org down ?

On 13 Apr 2010, at 15:22, Olivier wrote: Is it me or is svn.asterisk.org down ? issues. too -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

Re: [asterisk-users] Being attacked by an Amazon EC2 ...

On 12 Apr 2010, at 17:30, Tom Stordy-Allison wrote: Good article - might solve our problems for now: http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood He got the bots to stop by writing a ruby script that responds back to them with a SIP 200 OK. I'm going give it a

Re: [asterisk-users] cause 66 - Channel not implemented

On 12 Apr 2010, at 20:00, David Backeberg wrote: chan_carrierpigeon must be in asterisk-extras. I'll have to upgrade and check it out. Terrible latency, and seems susceptible to packet loss where shotguns are involved. S --

Re: [asterisk-users] asterisk start with php

On 2 Apr 2010, at 18:49, salaheddine elharit wrote: thank so much again for your response , i don't understand what shoud i do if you can please give me more information how to do in oreder to excute this script He's damned near written it for you. Try researching the terms he used, and

Re: [asterisk-users] SIP OPTIONS response from the peer is ignored - peer becomes UNREACHABLE

On 25 Mar 2010, at 10:18, Asterisk wrote: How is it possible that the peer becames UNREACHABLE eventhough Wireshark logged its proper response? Wireshark received it, doesn't mean Asterisk did. what does a sip debug in Asterisk show? S --

Re: [asterisk-users] new server install errors starting asterisk

On 25 Mar 2010, at 13:08, Ott Rose wrote: Can't find indications config file indications.conf. Thats the last line. Probably the problem... Amazing what reading instructions does... S -- _ -- Bandwidth and Colocation

Re: [asterisk-users] new server install errors starting asterisk

On 25 Mar 2010, at 14:02, Ott Rose wrote: well i followed the same directions i used like 3 weeks ago with 1.6.0 and didn't have any issue. Not sure what went wrong. That why i posted it. how can it work one time and not the next. Does the file exist? If not, then something is

  1   2   3   >