Re: [asterisk-users] asterisk server stress test

On 20 Aug 2015, at 11:12, Sevana Oy wrote: > Curious why didn’t you try AQuA to score > the quality? Using voice files for tests has more representation to my > opinion. Spot the salesman? ;) Steve-- _

Re: [asterisk-users] packages.digium.com

On 11 Mar 2015, at 17:53, Matthew Jordan wrote: > On Wed, Mar 11, 2015 at 10:28 AM, Steven Howes > wrote: >> Anyone know where it’s gone?.. Appears to have been down all day. > The hamsters should be running in their wheels again now. Cheers Matthew. Give them some food

[asterisk-users] packages.digium.com

Anyone know where it’s gone?.. Appears to have been down all day. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] (no subject)

On 9 Feb 2015, at 15:32, Francisco Leonardo Mota wrote: > Submission. > > Thanks, Uh, no problem?.. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introduct

Re: [asterisk-users] Subscribe event "ua-profile"

On 10 Nov 2014, at 13:01, Norman Laidla wrote: > Well, pants. It actually is causing a problem, because the phone doesn't use > any other methods to register to Asterisk. This is a bit of a big issue. What softphone is it? It sounds like rather odd behaviour. --

Re: [asterisk-users] Asterisk 13 stable?

> The Asterisk 13 is already stable for production environment? It’s only been out a couple of days - hard to make judgements just yet. But it is out of Beta so should be of reasonable quality - give it a try :) Steve -- _ -- B

Re: [asterisk-users] Asterisk Phone ( Telecom feature )

On 7 Oct 2014, at 09:24, Dania Asi wrote: > Kindly note that I asked about the capability of the phones and now I am > asking about the way I can do it to my client's phones, because he is asking > for a demonstration. Yet you’ve not even told us the phones in use. You can’t just expect a mailing

Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk

On 2 Sep 2014, at 10:38, Jonas Kellens wrote: > Then how can I let another Asterisk server know the custom reason of hangup ? > If it is not possible with custom SIP-header, then how ? As far as I know that’s going to require a source change. May not be the case with PJSIP though - not used tha

Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk

On 2 Sep 2014, at 09:03, Jonas Kellens wrote: > So just before hanging up, I add a custom SIP-header : > > exten => s,n,SIPAddHeader(X-My-Hangup: MaxChan) > exten => s,n,Hangup() SIPAddHeader only works for INVITE as far as I know. Steve-- __

Re: [asterisk-users] AMI & Elastix

On 18 Aug 2014, at 09:27, Усин Айбек wrote: > I have trouble with connection to AMI 1.1 wich enabled on Elastix > > "Asterisk Call Manager/1.1 > Action: Login Username: admin Secret: qweasd123 > > Response: Error > Message: Missing action in request" You are missing the newline characters.. Ac

Re: [asterisk-users] The plain old PBX functionality

On 8 Aug 2014, at 06:05, Gergo Csibra wrote: > back in the old analog telephony days there was "digital" PBX-es and > digital "system" phonesets. This phonesets have had many individual > illuminatable buttons connected with extensions. The PBX can show on > the buttons if some extension is ringin

Re: [asterisk-users] Unregistered ports on SPAxxxx

On 5 Aug 2014, at 17:10, Mike Diehl wrote: > All of my SPA112's are running 1.3.2(014). My SPA8000's are running 5.1.10. If you do firmware upgrade your 8000s, don’t go past 6.1.3 or it’ll go badly… Freezing and requiring power-cycle, clocks stopping (and showing minus figures!) and major str

Re: [asterisk-users] Attack on Sip server.

On 27 Jun 2014, at 15:37, Anurag Rana wrote: > There are lot of requests coming in and I am not able to stop it because I am > unable to detect the IP address. > I used wireshark to capture the packets. If you can capture the packet, surely you have the IP? If they intend to get the response t

Re: [asterisk-users] Elastix Architecture

On 2 May 2014, at 10:07, upendra wrote: > Am new to Elastix and wanted to try build new modules in the Elastix , so i > want to know how the PHP is running ?? as i see no Apache server inside ?? so > wanted to know how its running ? which server and architecture? This is not an Elastix mailing

Re: [asterisk-users] Numbers hackers call

On 26 Mar 2014, at 16:20, Michelle Dupuis wrote: > If this is to 972 area code then the next digits should be 0X or 0XX but they > are not. This differs from what I found documented for that area code - I > thought someone from the region might add to the discussion. Not sure if > this refl

Re: [asterisk-users] Numbers hackers call

On 26 Mar 2014, at 15:05, Michelle Dupuis wrote: > I see a lot of attempts by hackers to call 00972595301123​ or > 011972595115207​ or variations but that same 972595 is often present. > > Can someone break down that dial string with an explanation? The 011 look > like an overseas call (from

Re: [asterisk-users] IAXModem or T38Modem?

On 25 Mar 2014, at 15:00, Jeff LaCoursiere wrote: > On 03/24/2014 05:50 PM, Thorolf Godawa wrote: >> >> But your carrier has to support T38, when we began to evaluate this some >> years ago, this was not true for all. > Would you share the provider you are using? I have had almost zero luck so

Re: [asterisk-users] Spammer direct replying to those posting on the users list

On 25 Mar 2014, at 14:16, Digium's Asterisk Development Team wrote: > We apparently have a spam bot subscribed to the list and replying > *directly* to anyone who posts on the list. There’s plenty of people harvesting the list archives too, I get loads of spam about gateways etc :( S -- _

Re: [asterisk-users] Enterprise VoIP Trunk

Probably should post this to the asterisk-biz list. This is the non-commercial discussion list. Post to the -biz list and you’ll probably have loads of sales droids happy to help :) Steve -- _ -- Bandwidth and Colocation Provid

Re: [asterisk-users] SIP OPTIONS "storm"?

On 13 Feb 2014, at 18:10, Tim Nelson wrote: > I recently experienced an odd situation. I have an Asterisk 11.5.0 system > (Box A) with a SIP peering to another Asterisk 1.8.23.0 system (Box B). At > some point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box > A. I do have quali

Re: [asterisk-users] how can I get authenticate from my own server?

On 17 Jan 2014, at 02:18, Sean Darcy wrote: > I'm used to seeing fraudulent attempts to authenticate, But now I'm getting > them from the server itself. > > I have an asterisk server behind a firewalled router. The local subnet is > 10.10.10.0/24, the server is 10.10.10.100. > > Now I'm seein

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

On 14 Jan 2014, at 02:19, Patrick Lists 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. Steve -- _ -- Bandwidth and Colocation Pr

Re: [asterisk-users] Multiple IAX2 Trunks Load balancing

On 13 Dec 2013, at 07:48, Muhammad Usman wrote: > Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to load > balance incoming calls over IAX2 trunks. If any trunk goes down the calls > traffic will be shared with other available trunks. When it gets Up the > script is suppo

Re: [asterisk-users] Hack

On 18 Oct 2013, at 04:06, John T. Bittner wrote: > Today I was hacked but caught it very quickly. This is the weird part, they > hacked an IP Auth based account by simply knowing the account name. > > How is this possible? I am running Asterisk 11.5.0. Now it’s my fault I used > a dictionary ba

Re: [asterisk-users] RTP port ranges

On 13 Sep 2013, at 11:44, A J Stiles wrote: > In the Windows world, where you usually don't get the Source Code, you never > know what is running on your computer; in which case, you are never sure that > there isn't a daemon listening on a particular port number, so it is wise in > that case no

Re: [asterisk-users] Introducing Sippy Cup: SIPp Load Testing Made Easy

On 27 Aug 2013, at 15:34, Ben Klang wrote: > But what's REALLY useful is Sippy Cup's ability to dynamically generate PCAP > audio. If you've ever needed to drive an IVR from SIPp you're probably > familiar with the pains - it usually requires capturing an actual call, > isolating the RTP, and

Re: [asterisk-users] Cut off last character of EXTEN

On 20 Aug 2013, at 12:25, Pat Collins wrote: > Here ya go: > > 112233# use ${EXTEN:0:6}) > 123# use ${EXTEN:0:3}) > 123456789# use ${EXTEN:0:9}) I think 'variable length' implied 'unknown length'... S-- _ -- Bandwidth and Col

Re: [asterisk-users] Strange issues with newly rebooted machine

On 6 Aug 2013, at 19:28, Mike Diehl wrote: > We got it fixed! Our co-lo is in the process of doing a network > reconfiguration/relocation and had changed their MTU to 1400 during > the transition. Once we did the same, everything started to work. PMTU should take care of that. Are you blocking I

Re: [asterisk-users] Microsoft CRM Integration

On 16 Jul 2013, at 04:10, Klaverstyn, David C wrote: > I’m hoping someone can recommend a method to integrate Microsoft CRM with > Asterisk. Preferably an open source product otherwise a commercial product. Hi, You've not said what you're trying to integrate... Creating tasks for calls, contac

Re: [asterisk-users] SIP. Call-limit dialstatus

On 3 Jul 2013, at 12:28, I.Pavlov wrote: > [2013-07-03 15:22:27] NOTICE[29728]: chan_sip.c:6003 update_call_counter: > Call to peer '0014' rejected due to usage limit of 1 > -- Couldn't call 0014 > == Everyone is busy/congested at this time (0:0/0/0) > -- Executing [0014@sub_pbxdialco:50

Re: [asterisk-users] Jabber

On 23 May 2013, at 10:49, bilal ghayyad wrote: > Facebook and Whatsapp sort-of support XMPP, so we can use Jabber to > communicate with them. But, how much jabber channel in asterisk is stable and > updated? You can find out the support status from menuselect https://wiki.asterisk.org/wiki/disp

Re: [asterisk-users] Asterisk 1.6 + Cisco AS5300

On 7 Mar 2013, at 10:12, Mickael Monsieur wrote: > Do you have an explanation? Put a SIP debug on and we may be able to find one.. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk?

Re: [asterisk-users] POKE from command line

On 26 Feb 2013, at 16:52, Gary Carr wrote: > Is it possible to issue the POKE to a end point from the CLI? Our asterisk > servers is not seeing some end points drop off and I would like to create a > script to manually check end points. http://www.geekinter.net/iaxping.txt May be of use to you.

Re: [asterisk-users] Dialplan / check / tool

On 18 Feb 2013, at 17:03, Christopher Harrington wrote: > On Mon, Feb 18, 2013 at 10:54 AM, Steve Edwards > wrote: > ) If the AGI is [...] a compiled executable (C, Fortran, Cobol, assembler, > etc.) > I'd like to see an AGI written using Fortran or Cobol. Don't tempt me ;) S-- _

Re: [asterisk-users] RPM updates

On 6 Feb 2013, at 20:06, Rusty Newton wrote: > - Original Message - >> From: "Steven Howes" >> On 28 Jan 2013, at 13:55, Steven Howes wrote: >>> Who do I need to poke to get the yum repository / RPM files >>> updated? The dahdi RPMs are not up

Re: [asterisk-users] RPM updates

On 28 Jan 2013, at 13:55, Steven Howes wrote: > Who do I need to poke to get the yum repository / RPM files updated? The > dahdi RPMs are not up to date with the CentOS kernel versions any more, it's > making doing an installation a bit tricky due to dependancies, I'd rath

Re: [asterisk-users] CallerID external call after Attended Transfer

On 4 Feb 2013, at 13:45, Jonas Kellens wrote: > The IP-phones in this case are Yealink T32G. > > What setting is needed in this IP-phone ? Quick google doesn't turn up any results. Handsets probably dont support it. Steve-- _ --

Re: [asterisk-users] CallerID external call after Attended Transfer

On 4 Feb 2013, at 12:53, Jonas Kellens wrote: > I call with my cellphone to our public telephone number > Our receptionist answers the incoming call and does an attended transfer to > my colleague ( A ) > Colleague answers and the receptionist tells him that I am on the other side. > Receptionist

[asterisk-users] RPM updates

Hi All, Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. Cheers

Re: [asterisk-users] DIDForSale spam

On 10 Jan 2013, at 02:10, Jai Rangi wrote: > I have removed yours right away. > > Yes, I agree, But just like any company we have purchased/collected email > from different source. Also just like any company we are not perfect, we make > mistakes. Then buy your addresses from different source

Re: [asterisk-users] Verizon SIP "trunking" Field Trial

On 3 Jan 2013, at 15:13, Michael L. Young wrote: > So, I am asking the community for any input. I have read on here and seen on > IRC that some in the community are successfully using Asterisk with Verizon > SIP. Verizon was going to check and see if they have any notes about that > and those

Re: [asterisk-users] Top Posting

On 2 Jan 2013, at 15:54, Eric Wieling wrote: > On Sun, Dec 30, 2012 at 2:54 PM, Benny Amorsen > wrote: >> There is a sanction. People like me will score top posters lower and >> soon not see their posts at all. > I'm the opposite. I'm likely not to scroll down 10 pages to see the comments

Re: [asterisk-users] Top Posting

We should all top *AND* bottom post! On 31 Dec 2012, at 06:03, isr...@gmail.com wrote: > Just my pitch in to post > From a blackberry you can only top post there is no way of bottom posting > So if I would have to wait to get to a computer to bottom post I would just > never answer We should al

Re: [asterisk-users] ODBC Connection Problem

On 10 Dec 2012, at 16:13, Christopher Harrington wrote: > On Mon, Dec 10, 2012 at 5:23 AM, Chandrakant Solanki > wrote: > Password= c3podb@2012 > > In case you didn't realize you were sending this out publicly to a publicly > archived and searchable list, you might want to change that p

Re: [asterisk-users] Why all the 401 Unauthorized

Hi, SIP registrations typically try to register, are them prompted for a password (via a 401 message) it then sends a new request with authentication . This is normal. Steve On 23 Oct 2012, at 13:26, Jerry Geis wrote: > I have a connection between two asterisk boxes, both running 1.4.43 > >

Re: [asterisk-users] realtime sip peers status

On 22 Oct 2012, at 15:21, Ishfaq Malik wrote: > On Mon, 2012-10-22 at 19:17 +0500, Control Oye wrote: >> I have successfully setup Asterisk realtime. Now I can create >> extensions dynamically. But when I put this command on cli mode >> >> sip show peers >> >> it returns no result. >> >> can any

Re: [asterisk-users] asterisk as IVR using 3g usb modem

On 18 Oct 2012, at 16:50, Mitul Limbani wrote: > U would have to write a dahdi module for this 3G modem to help asterisk > understand it as standard gsm channel. > Look up chan_datacard (i think that's what it's called from memory). Steve --

Re: [asterisk-users] using analog phones

On 20 Aug 2012, at 01:37, Noam Birnbaum wrote: > A client wants to keep their old Inter-Tel KTS analog phones for budget > reasons. Two questions: > > 1. How could they use these with FreePBX? You'd need an ATA or an analogue card. And how well they work, depends on if they are 'true' analogue,

Re: [asterisk-users] HP DL360 G5 better than HP DL360 G7 ?

On 1 Aug 2012, at 12:39, D Tucny wrote: > HP refuse to let you put your data at risk, refusing to activate write > caching without a charged battery attached or NV cache. > > I personally would like to have the option to override things like this at my > own risk, but, HP don't give you that op

Re: [asterisk-users] Problem provisioning Cisco SPA303

On 1 Aug 2012, at 07:05, Support wrote: > I've looked on Cisco's website and Googled around, but I can not find a true > example of a provisioning file for this device. Anything I could find would > be > enough for me to make a template. Download the SPC tool from the Cisco site. It includes t

Re: [asterisk-users] Digium IP Phones D40

On 25 Jun 2012, at 16:58, Jeff LaCoursiere wrote: > Actually we get that complaint a lot too (Polycom ring volume). We > typically install in hotel environments, and in their back office the > environment can be noisy, as well as in their restaurants. > > I imagine in a typical office environment

[asterisk-users] Fwd: Flashphoner

Thought this deserved a name and shame! ;) Steve Begin forwarded message: > From: Pavel Ismailov > Date: 27 April 2012 06:58:07 GMT+01:00 > To: steve-li...@geekinter.net > Subject: Flashphoner > > Hello! > > My name is Pavel Ismailov > and I`m CEO of www.flashphoner.com project. > > We not

Re: [asterisk-users] Company info

Can't tell if this is a transparent attempt at advertising, or...? S On 19 Apr 2012, at 22:09, Josué Conti wrote: > This is your website: > > http://www.convergia.com/ > > Thanks in advanced for any informations. > > Best Regards > > Josue > > Em 19 de abril de 2012 17:11, Josué Conti escr

Re: [asterisk-users] Voicemail crashs asterisk

On 3 Apr 2012, at 16:42, Vik Killa wrote: > #disasterisk fail > > #freeswitch win #unhelpful comment S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] unsubscribe

On 30 Mar 2012, at 10:04, Sean McMaster wrote: > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users It's not tricky.. Really.. It's on the bottom of every email.-- ___

Re: [asterisk-users] concurrent channels limit

On 30 Mar 2012, at 10:14, Syco wrote: > Finally the problem is: I cannot manage more than 80 concurrent calls. What happens on the 81st call?.. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

Re: [asterisk-users] use of Read cmd with AGI

On 2 Mar 2012, at 11:35, Kamlesh Kumar wrote: > $agi = new AGI(); > $agi-> exec('Background','press_one0&press_two0&press_zero0'); > $agi-> exec('Read','NUMBER,,1,3'); > $agi-> verbose ("You have entered".$NUMBER); You need to use AGI to read the Asterisk variable.. Asterisk variables don't magi

Re: [asterisk-users] India Pune Pri call problem

On 13 Feb 2012, at 12:06, virendra bhati wrote: > You can't set callerid for outgoing calls in case of PRI. Why not? Every PRI I have used supported it. Is this a carrier-specific thing? S -- _ -- Bandwidth and Colocation Provide

Re: [asterisk-users] Stumped about adding a semi-colon to a variable

On 11 Feb 2012, at 13:41, Kevin P. Fleming wrote: > At this time, there's no way to do it directly in the dialplan In extensions.conf [globals] SEMICOLON=; Then use ${SEMICOLON} S-- _ -- Bandwidth and Colocation Provided by htt

Re: [asterisk-users] Major changes between 1.4/1.6.1.8/10?

On 9 Feb 2012, at 11:08, Gilles wrote: > Does someone of a good site/blog that keeps track of new releases of > Asterisk, and explains what the major changes/features when they do > occur? Why not just use the latest version?.. S -- ___

Re: [asterisk-users] Major changes between 1.4/1.6.1.8/10?

On 7 Feb 2012, at 14:27, Gilles wrote: > On Mon, 06 Feb 2012 10:24:42 -0600 (CST), Richard Mudgett > wrote: >> The UPGRADE.txt and CHANGES files do just that. They have been a part >> of the Asterisk source files for a long time. > > Thanks for the info. The problem is that the ChangeLog files >

Re: [asterisk-users] Securing Asterisk

On 27 Jul 2011, at 17:11, CDR wrote: > This is turning into a political issue such as the one in Washington > and the impending default on US debt. No, YOU are turning this into a political discussion. > The point is that a minor change > in the code would have a dramatic effect on security, and

[asterisk-users] Asterisk 1.6.2.19 RPM

Hi All, Asterisk 1.6.2.19 was released on the 28th, does anyone know if there a timescale for this reaching the RPM repository? We're badly affected by a bug in previous versions that has only recently become apparent to us. It's in a situation where rebuilding from source isn't too practical s

Re: [asterisk-users] Cannot figure out pound key in qwerty keyboard

On 30 Jun 2011, at 16:14, Danny Nicholas wrote: > #1 . # - Shift-3 should always work Not in the UK, or other countries that have different layouts for symbols.. S -- _ -- Bandwidth and Colocation Provided by http://www.api-dig

Re: [asterisk-users] Re : Re : Re : Re : Direct RTP with Asterisk

On 20 Jun 2011, at 16:33, Sagbo Romaric wrote: > Any client behind his NAT can talk with another behind his NAT. Still not possible.. The internet doesn't really work like that. SIP even more so. S-- _ -- Bandwidth and Colocatio

Re: [asterisk-users] VOICEMAIL CONFIGURATION

On 15 Jun 2011, at 11:20, mahesh katta wrote: > i DID SOME VOICE MAIL CONFIGURATION. SO HOW CAN YOU RETRIVE THAT VOICEMAIL. > WHEN I RETRIVE THE VOCIE MAIL ITS NOT GETTING ANY MSG A lot of filesystems are case sensitive. Maybe you wrote your configuration in caps? This would also explain why you

Re: [asterisk-users] Voxbone numbers

On 3 Jun 2011, at 11:57, devr devr wrote: > My query now is willl all voxbone numbers show up as the operator as Voxbone > SA as above. I wanted to find out who the service provider is on some > numbers, I suspected the service to be voxbone but the operator shows as > other companies. > > My

Re: [asterisk-users] Question about "null routing" calls to DIDs we don't handle

On 1 Jun 2011, at 22:50, Jesse Thompson wrote: > We are managing an Asterisk installation for residential VOIP service, and we > are having a problem where all inbound calls to DIDs which are assigned to us > by our wholesaler but not yet assigned to a downstream customer get caught in > a routi

Re: [asterisk-users] UK English sounds packs

On 26 May 2011, at 14:09, Ishfaq Malik wrote: > Does anyone know if there are any free UK accented English sounds packs? http://www.tel.net Not perfect, but damned near :) S -- _ -- Bandwidth and Colocation Provided by http://w

Re: [asterisk-users] receive faxes

On 4 May 2011, at 15:01, vip killa wrote: > screw that i just got hylafax to work with IAXMODEM...i refuse to pay > digium a dime... supposed to be open-source right? There is so much wrong with that sentence, I don't know where to start. On 4 May 2011, at 16:02, vip killa wrote: > Honestly

Re: [asterisk-users] Asterisk, bicolor BLF and DEVSTATE

On 3 May 2011, at 13:34, Olivier wrote: > It seems that Asterisk DEVICE_STATE function can't be mapped with this > bicolor feature. > So how could this 4-states BLF be implemented ? > Any suggestion ? The handset is what maps the colours to the states. You can't send a colour in SIP. You could

Re: [asterisk-users] Asterisk Export Fax from Wave file

On 21 Apr 2011, at 13:46, A J Stiles wrote: > You *might* be able to recover the document, *if and only if* the recording > quality is high enough. Easiest way to try it is to call up a fax machine > (either an actual real one, or a copy of Hylafax) from Asterisk and play the > wav file down

Re: [asterisk-users] Call Center Reporting

On 18 Apr 2011, at 11:06, bilal ghayyad wrote: > I am using Asterisk for Call Center (so agents login, logout, ready, not > ready, ... etc). To be able to have a good call center reporting, on what I > have to depend? On the CDR of Asterisk or there is another way? > > Is there a good open sourc

Re: [asterisk-users] voicemail odbc "Length is ....."

On 11 Apr 2011, at 15:28, vip killa wrote: > I'm using voicemail ODBC with Asterisk 1.6.2.17.2. > Why do I see "Length is 186545" or something similar but a different number > in Asterisk CLI everytime someone leaves a message? Because not all messages are the same length. I'd guess it's length

Re: [asterisk-users] changing port 5060 to 5061

On 11 Apr 2011, at 10:03, darin iv wrote: > please send me the ways to change asterisk port from 5060 to 5061 i > need to configure it because we are already using 5060 port in router > then we cant use it again we have to configure other sip server so > please suggest me a way.

Re: [asterisk-users] asterisk SIP MESSAGE method support

On 7 Apr 2011, at 14:32, Deka, Rajib IN MAA SL wrote: > Is the following is the link for getting the source, > http://svn.asterisk.org/svn/asterisk/trunk/ Please try not to reply to the entire digest.. S -- _ -- Bandwidth and Co

Re: [asterisk-users] voicemail call back loop

On 6 Apr 2011, at 17:46, vip killa wrote: > I have "externnotify = /var/lib/asterisk/agi-bin/vm_notify.pl" so that when > someone is left a voicemail it will call the person's mobile phone and prompt > them with the new message. The perl script simply originates a call to a > persons mobile phon

Re: [asterisk-users] Call recording - methodology

On 6 Apr 2011, at 11:54, Silver Thorne wrote: > Does anyone know of any opensource or otherwise solutions out there that I > can try out? Asterisk. Google it. If you're too lazy, Google MixMonitor. If you're too lazy for that: http://www.voip-info.org/wiki/view/MixMonitor S -- ___

Re: [asterisk-users] is downloads.asterisk.org down?

On 31 Mar 2011, at 13:52, Sebastian wrote: > http://www.downforeveryoneorjustme.com/downloads.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introduct

Re: [asterisk-users] asterisk and fail2ban

On 28 Mar 2011, at 14:19, vip killa wrote: > Yes I followed directions on that page > Running Asterisk 1.6.1.22, anybody else experiencing this? How often does fail2ban check the logs? It can only block that often, so if more attempts happen in that time period it can't do anything until it knows

Re: [asterisk-users] Fwd: asking for some help

On 24 Mar 2011, at 16:46, tahar .H wrote: > so plz is there any one who can give me a puch to learn this extraordinary > Asterisk plz(video things will be better :)) Learn to ask questions. Learn to read books. Learn to use google. S -- ___

Re: [asterisk-users] What is the most stable version of asterisk?

On 24 Mar 2011, at 16:38, Gordon Henderson wrote: > 1.2 has been the most stable version for me. > > Same setups with 1.4 +DAHDI has never been as stable with random crashes and > re-starts - however they're not predictable and sometimes months apart. I had > one instance of 1.2 run for over a y

Re: [asterisk-users] Asterisk using as a SIP client

On 23 Mar 2011, at 10:40, Nikhil wrote: > I am planning to use asterisk as a IP phone(Porting asterisk into a hardware). Interesting.. > Is there any limitations if I use asterisk as a SIP client?,and asterisk has > any advantages if use like this? It's not really designed as a SIP client. It's

Re: [asterisk-users] Multi-Tenant Hosted PBX system with Reseller functionality

On 22 Mar 2011, at 01:09, Outback Dingo wrote: > Even worse... now it smells of a scam At least their website isn't hideous... Oh..wait.. ;) S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to A

Re: [asterisk-users] SIPAddHeader not working

On 15 Mar 2011, at 15:21, Jonas Kellens wrote: > On 03/15/2011 12:39 PM, Steven Howes wrote: >> >> On 15 Mar 2011, at 11:30, Jonas Kellens wrote: >>> On 03/15/2011 12:24 PM, Steven Howes wrote: >>>> >>>> On 15 Mar 2011, at 09:08, Jonas Kellens wrot

Re: [asterisk-users] SIPAddHeader not working

On 15 Mar 2011, at 11:30, Jonas Kellens wrote: > On 03/15/2011 12:24 PM, Steven Howes wrote: >> >> On 15 Mar 2011, at 09:08, Jonas Kellens wrote: >>> I also notice the presence of a "Remote-Party-ID" SIPheader... Where does >>> this come from ?! Not

Re: [asterisk-users] SIPAddHeader not working

On 15 Mar 2011, at 09:08, Jonas Kellens wrote: > I also notice the presence of a "Remote-Party-ID" SIPheader... Where does > this come from ?! Not from my dialplan... sendrpid in your sip.conf Steve-- _ -- Bandwidth and Colocat

Re: [asterisk-users] Asterisk 1.8 paging with ploycom

On 14 Mar 2011, at 16:24, satish patel wrote: > I test page application and it works but i am worried about i have 200 SIP > phone. Do you think asterisk page application can handle that number of page > ? Do they support multicast? S-- _

Re: [asterisk-users] SIPAddHeader not working

On 14 Mar 2011, at 15:58, Jonas Kellens wrote: > dialplan : > > exten => 67121212,1,NoOp() > exten => 67121212,n,Set(CALLERID(all)="3259" <3259>) > exten => 67121212,n,SIPAddHeader(P-Preferred-Identity: > ) > exten => 67121212,n,SIPAddHeader(Privacy: id) > exten => 67121212,n,Dial(SIP/32

Re: [asterisk-users] Testing from where number is...

On 3 Mar 2011, at 20:53, Danny Nicholas wrote: > Not having an in-depth knowledge of how EU numbering works Sadly there is no 'EU numbering'. Europe isn't a country, thus doesn't share any dial plan. There appears to be some tendency towards having a '7' at the front of a mobile number, but it's

Re: [asterisk-users] asterisk security....again

On 28 Feb 2011, at 10:33, Rizwan Hisham wrote: > The problem I have been experiencing since last month is that some of my > customers are getting calls with "Asterisk " caller id. Most of them > in the middle of the night. And my asterisk server has no record of these > calls. The customers were

Re: [asterisk-users] Regarding Asterisk

On 17 Feb 2011, at 17:52, Fred Posner wrote: > Awesome. Any institution that issues a masters to someone who asks a > question to a mail list on open-source software must be held in high > regard. Careful! You might end up with a Masters in Debating. S --

Re: [asterisk-users] Asterisk Using as a SIP Client

On 17 Feb 2011, at 10:04, Nikhil wrote: > Do I need to modify chan_phone application to make it works or it is > available in net. Why not use a proper sip client? S -- _ -- Bandwidth and Colocation Provided by http://www.api-di

Re: [asterisk-users] Asterisk and Cisco Call Manager Express (CME)

On 24 Oct 2008, at 03:57, David Gibbons wrote: > Dare I ask why you want to do this? Cheaper than buying an AIM-CUE? And certainly more flexible. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To U

Re: [asterisk-users] : Parking Issue

On 22 Oct 2008, at 20:29, Craig Van Ham wrote: > > HI all, > > This appears to be the same message you posted earlier. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update opt

Re: [asterisk-users] app_confcall build issues

On 16 Oct 2008, at 14:57, jonathan augenstine wrote: > I am trying to build app_confcall and it is failing. Are there > known build issues with this module. I am running Asterisk 1.6.0- > beta9. Ah yes. 'failing'. I bet that is all it says eh? its not like compilers give descriptive errors

Re: [asterisk-users] Call files

On 14 Oct 2008, at 18:05, Christian Victor wrote: > Steven Howes schrieb: >> Have created a system that involves using call files in the outgoing >> spool folder. On some occasions it retries which is fine is there >> any way to view calls waiting retries from the CLI? U

[asterisk-users] Call files

Hi All, Have created a system that involves using call files in the outgoing spool folder. On some occasions it retries which is fine is there any way to view calls waiting retries from the CLI? Using 1.4 btw. Have googled to no avail (although it is near the end of the day so I might

Re: [asterisk-users] Asterisk voicemail

On 14 Oct 2008, at 11:00, Chris Rowson wrote: >> Hi folks, >> >> I'm working on a solution using the Asterisk voicemail component and >> wondered if anyone knew the answer to this question please? >> >> I understand that Asterisk saves voicemail to >> /var/spool/asterisk/voicemail///INBOX/ but I >

Re: [asterisk-users] Tribox

Hi triXbox.org can answer these questions. Google may also give a balanced view. But yes, i can assure you, people are using Trixbox from Fonality. Steve On 6 Oct 2008, at 10:24, broadband Voice wrote: > Anyone using Tribox from Fonality. I understand its open source and > free. Can I use

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