[asterisk-users] taskprocessor.c: The 'sorcery/contact-00000015' task processor queue reached 1500 scheduled tasks.

Running 15.1.2. I have four devices transitioned to use pjsip. After about 1-2 days of uptime, psjip stops accepting registrations, and the messages log contains the entry as per the subject. At any given time, "pjsip show contacts" only shows the four devices. Could someone point me to a fix,

[asterisk-users] PJSIP console messages with Zoiper

I'm running Asterisk 15.1.0 and in the process of converting my various SIP endpoints to use PJSIP. My Zoiper client causes the messages quoted below to show up on the CLI once per minute. Things seem to work OK, but I am curious because there seems to be no way to suppress the messages, and ther

[asterisk-users] Any way to stop Playtones(dial) when the user presses a key, emulating a CO's behavior?

The subject says it all. I'm betting there's a way to do it, but so far I haven't found the dialplan runestone via web searching. Thanks. b. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to As

[asterisk-users] Exceptionally long queue length queuing . . . .

I wonder if anyone out there has a perspective on this. There are a welter of tickets out there on the matter, most of them closed. This problem began for me over a year ago, and continues up to the latest versions I've installed (1.6.2.13). It happens randomly, and the suggestion on one of th

Re: [asterisk-users] (no subject)

Alex Balashov wrote: ) How about rejecting emails that don't have a subject? That is an excellent idea. If a person doesn't have enough clue to use a subject, then we're really just feeding the beast when we indulge the question with an answer. And the archived version of that question/an

Re: [asterisk-users] need * consultant in houston area

Could you all please take this COMMERCIAL discussion to the -biz list? Thanks. b. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.

Re: [asterisk-users] Gemeinschaft released

Philipp Kempgen wrote: >> The cluster capability is very interesting. > > Exactly. Although I must admit wo don't really have much > documentation on how to install a "Gemeinschaft cluster". > Is there anything documented in English? Philipp's mail doesn't say what the product does, and the we

Re: [asterisk-users] Asterisk content @ OSCON 2008?

Martin Smith wrote: > Hey folks, > > Is anyone working on Asterisk (or other) presentation proposals for > OSCON 2008 in Portland, OR? Here's the link, in case: > http://en.oreilly.com/oscon2008/public/cfp/13 > I'm working on a proposal to do a tutorial there on Asterisk. I have done such the

Re: [asterisk-users] Realtime: Should I say or should I go (now) ?

Terry Wilson wrote: > > And a free hint: if you are going to have to do anything that > resembles number porting, swapping extensions, etc.--don't use > extensions/phone numbers as SIP usernames. You have to regenerate > config files, etc. Make your SIP usernames meaningless and use > f

Re: [asterisk-users] Asterisk 1.4: encryption support

Russell Bryant wrote: > Alejandro Cabrera Obed wrote: > >>Dear all, I have Asterisk 1.4.13 and I need to use encryption among >>Asterisk and my SIP users, and with the RTP data interchanged among >>users. I prefer the use of ZRTP/SRTP because we use Twinkle and >>X-Lite/Zfone as our voip clients a

Re: [asterisk-users] Realtime on Asterisk 1.2.24

Tilghman Lesher wrote: > > > I *STRONGLY* recommend that you do NOT use realtime extensions. If you > want a dynamic dialplan, the correct way to do it is to segregate your logic > and your data (via something like func_odbc), not to stick all of your logic > into a database. > Should this be

Re: [asterisk-users] Asterisk on USB Flash?

shadowym wrote: > Size/Speed/write cycles have gone way up, price has gone way down. More > common than CompactFlash and no need for an adapter. So is it feasible to > run an Asterisk server on something like this? With a MTBF of 1million > write cycles coupled with dynamic wear management on a

Re: [asterisk-users] phone as control interface (was 99bottlesof beer)

Jon Pounder wrote: > Quoting John Faubion <[EMAIL PROTECTED]>: > > >>>Does anyone know of such a device that I can use over a network? It would >>>be a pain to run a USB cable. I am thinking of devices that are like: >> >>I think your missing the key feature of these devices, UPB/X10. UPB and X1

Re: [asterisk-users] What's the deal with ATAcomm?

Andrew Joakimsen wrote: > That's horrible. I don't buy too many IP phones these days, but can > anyone suggest a place better than the scumbags at VoIP supply? Reading this raised my eyebrows. I'm not sure what the content will do for your readers regarding the company; I am absolutely sure what

Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)

shadowym wrote: > Yes thank you for reminding me it is open source. Thank you for reminding > me people can write their own code for it. > > I'll get right on rewriting the entire sip code. Should only take me a few > hours. Including a couple hours to learn how to write c code. How hard can >

Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)

Matthew Fredrickson wrote: > shadowym wrote: > >>Maybe his comments were taken out of context as they don't have the whole >>interview posted. Why is he talking about queue games, Biologicall and >>other extremely niche crap when there are huge holes in the basic offering >>(SLA and SCA)? > >

Re: [asterisk-users] where is 1.4.12?

Matt wrote: > Absolutely not! However, if someone gave me a free flight, but the > plane went down 3 out of the 5 times it took off, yes I would :) > Then, if the makes of the plane released a new version where they > "fixed" the problem, but now instead of going down because the motors > shut off

Re: [asterisk-users] Teliax Quality of Service

Stephen Bosch wrote: > > PSTN service still sets the standard. > With infrastructure paid for under a gracious guaranteed-profit monopoly by ratepayers, now being used as a weapon to stifle competition from VoIP, cable, and other emerging technologies. b. -- This message has been scanned f

Re: [asterisk-users] Teliax Quality of Service

Stephen Bosch wrote: > Douglas Garstang wrote: > >>I confused by this. Don't ITSP's have redundancy? Don't they have >>multiple edge systems for accepting incoming calls? Don't their multiple >>edge systems have multiple interfaces, connected to multiple subnets, >>via multiple switches? And, don'

Re: [asterisk-users] asterisk is not sip proxy

Alex Balashov wrote: > The distinction between a back-to-back user agent and a proxy is a rather > formal one; (. . . . ) I suggest the essence of this mail be distilled and put into a FAQ somewhere, if it isn't already. Thanks. b. -- This message has been scanned for viruses and dangerou

Re: [asterisk-users] Caller ID Spoofing to be banned in the USA

J. Oquendo wrote: > Stephen Bosch wrote: > >> I ask that you treat people respectfully on the list. The poster has a >> valid point and does not deserve that kind of response. >> >> It's possible to disagree and still be civil, and I've no doubt you're >> able to do it. >> >> Thanks, >> >> > >

"Me Too" mails DOSsing the list (was: Re: [asterisk-users] Voip-info.org)

Compnet Bobby wrote: Same in southern cali! Folks, at the risk of sounding mean, once there are a couple of "Me Too" emails on a given topic, everyone who reads the list is clued in, and we don't really need another hundred or so one- or two-word emails adding to the already-bad load of tra

Re: [asterisk-users] Outside lines are just not happening...

Stephen Bosch wrote: And what do you do when they say: "We have a modern, relatively-new switch for which that sort of feature change is a trivial click on a GUI checkbox. However, we do not have any tariffed requirement to provide disconnect supervision. So we won't do it for you. Goodbye."

Re: [asterisk-users] Outside lines are just not happening...

Stephen Bosch wrote: If you get dumbfounded responses ask to speak to someone in the programming group (unless they are a tiny little phone company, they will have one). If you open a ticket, it usually means they will escalate the problem, even if the agent you are speaking with has no idea wha

Re: Now way OT: [asterisk-users] Need some help with a very simple Queestion..

Stephen Bosch wrote: Is Marmite also available in Ontario, or only Out West? As far as I know, Marmite is available all across this land, from sea to sea to sea. Three cheers for Marmite. IMO most Americans have never even *heard* of Marmite, much less tasted it. And it's quite a hoot t

Re: [asterisk-users] Improving Asterisk's DNS support

Kristian Kielhofner wrote: The intent of my original message was to try to start a discussion on how we can fix a REAL, KNOWN problem with Asterisk to make it better. I'm not sure what the exact problems are (or even the specific symptoms in all cases), but I am willing to offer anything tha

Re: [asterisk-users] wireless desktop phones

Jordan Novak wrote: Okay, I get it. I still have a problem though. I have no way to wire 30% of these end-points. P{hysically impossible. They do have cat3 twisted pair to each phone. But of course they want IP. Are there any adpaters that will give me just enough bandwidth to get it done. The

Re: [asterisk-users] Need feedback on vitelity

Mail list wrote: Hello Anyone here uses Vitelity as voip provider ? Their pplans looks good but i need some feedback from existing customers if any here . I would like to express an opposite opinion. I have two accounts with them with lots of DIDs. Everything works fine, and they have

Re: [asterisk-users] A request for your input.

Stephen Bosch wrote: Bill Hackensack wrote: On 3/22/07, [EMAIL PROTECTED] * <[EMAIL PROTECTED] > wrote: Hello P.S The program that I am using is open source, of course ( www.phpsurveyor.org )! What par

Re: [asterisk-users] voip-info.org status update

shadowym wrote: If I can't be confident enough in an important source of information like this then I can't be confident enough to provide an Asterisk solution to businesses. That's the way I see it. Yea, it's a wiki but it's the best source of info out there. Suggestion: switch to somet

Re: [asterisk-users] asterisk on mini-itx

Chris Mason (Lists) wrote: Here's how I do it. Buy complete fanless system flash card ready unit with four ethernet interfaces: http://www.ibt.ca/v2/items/fwa7204/index.html It is very small, in an aluminium extrusion case, very robust. What kind of money are those things? There doesn't see

Re: [asterisk-users] REALTIME and VIEW ENTIRE DIALPLAN

Davis Sylvester III wrote: Brian Capouch wrote: Davis Sylvester III wrote: Is there a way to view the entire dialplan when using Realtime? I use Realtime and MySQL connector. If you mean the contents of .conf-file based merged with whatever the Realtime engine is supplying, I don&#

Re: [asterisk-users] REALTIME and VIEW ENTIRE DIALPLAN

Davis Sylvester III wrote: Is there a way to view the entire dialplan when using Realtime? I use Realtime and MySQL connector. If you mean the contents of .conf-file based merged with whatever the Realtime engine is supplying, I don't think there's a way of seeing both together. But you ca

Re: [asterisk-users] Re: Asterisk Realtime

Mike Hammett wrote: [Mar 7 14:12:37] DEBUG[4380]: res_config_mysql.c:623 mysql_reconnect: MySQL RealTime: Successfully connected to database. [Mar 7 14:12:37] NOTICE[4380]: config.c:1174 ast_config_engine_register: Registered Config Engine mysql [Mar 7 14:12:37] NOTICE[4380]: config.c:1174 as

Re: [asterisk-users] Asterisk Realtime

Mike Hammett wrote: Could someone provide some steps for troubleshooting Realtime? I can’t see any signs that it’s working. I followed and double-checked a few different guides around the net, but haven’t been able to figure it out. You don't say which version you're running. I *think* the

Re: [asterisk-users] Sending SMSa

Michiel van Baak wrote: On 17:53, Sun 25 Feb 07, Brian Capouch wrote: Supa wrote: Try this: http://www.bayhamsystems.com/asterisk.html Works for me just fine, and it is very easy to get up and running, even with older version 1.2.3 Anyone out there running it against 1.4.0? It built

Re: [asterisk-users] Sending SMS

Supa wrote: Try this: http://www.bayhamsystems.com/asterisk.html Works for me just fine, and it is very easy to get up and running, even with older version 1.2.3 Anyone out there running it against 1.4.0? It built just fine for me, but then it crashes the server when I try to run it. Cra

Re: [asterisk-users] cisco sip firmware update for cisco 7970

Tim Connolly wrote: You can buy smartnet on a single phone for something like $8 a year. This will get you in legally. Any idea about how specifically to get such a contract? It is rumored to be pretty tricky. B. -- This message has been scanned for viruses and dangerous content by MailS

Re: [asterisk-users] CWI, call-limit and incominglimit

Pavel Jezek wrote: it can be also usefull to use 'limitonpeer' option in sip.conf with this, it would not be needed to define separate type=user and type=peer for each phone, instead define one type=friend and apply limitonpeers=yes ;limitonpeers=no; Apply all call limits (

Re: [asterisk-users] Problem with busydetect and cell phones

Stephen Bosch wrote: And then he hung up on me. ...wow. This society is doomed. Actually, it isn't so much society as the legacy telcos. But unfortunately, they've been pretty smart about using the billions that they've stolen from us over the years: they use a lot of it to line the p

Re: [asterisk-users] Problem with busydetect and cell phones

Stephen Bosch wrote: Have you tried calling AT&T and asking for call disconnect supervision? I realise that this can be a thankless and tedious endeavour, but it IS worth trying. There are almost no commercial switches that don't support this; it's a matter of activating it for the specific cir

Re: [asterisk-users] PRI Call Start

Stephen Bosch wrote: And use a different Wiki engine! Augh! (Mediawiki, anyone?) Who runs voip-info.org? I'll bet if you volunteered to take it over, the folks who run it would gladly let you have it And I'd further bet they'd gladly let you run whichever Wiki software you want!! Othe

Re: [asterisk-users] List problem handling HTML E-mails?

Yuan Liu wrote: My multiple postings to this list this morning got garbled in http://lists.digium.com/pipermail/asterisk-users/, and don't come back from list. (e.g., http://lists.digium.com/pipermail/asterisk-users/2007-February/179315.html) I thought it was Hotmail, so I saved one outgoing

Re: [asterisk-users] realtime extensions, labels

Julian Lyndon-Smith wrote: I cannot seem to find any reference to labels in realtime extensions - using 1.4. I've googled until my eyes have bled, and also scoured voip-info.org. Is there anything that helps me here ? You have to have numbered priorities with realtime. This is because (as

Re: [asterisk-users] Re: Best inexpensive home office router forVoIP (QoS with maybe PoE)

Jon Pounder wrote: Buzzwrong answer! Don't answer on things you have no idea. and stop providing bad information. you should take your own advice - an acre is 200ft x 200ft - what idiot would pay a consultant $7000 to tell them they need one access point in the middle. This is g

Re: [asterisk-users] Some queries on g729 license.

Al Bochter wrote: Matthew I agree. I only know what I have told by others so I do need this input I have been told that Digum G729 is a big pain the the butt to get working with Asterisk and it is very hard on the CPU Keep in mind I have never used any Ver. of G 729 So tell me what you thin

Re: [asterisk-users] How accurate is show translation?

Steve Underwood wrote: Paul Hales wrote: When you built Asterisk, it must have refused to build the ilbc codec - I have never seen an Asterisk box that could not transcode ilbc, in over 3 years of working with Asterisk. Most versions of embedded Asterisk will choke unto unusable if they are

Re: [asterisk-users] Need quality toll free 800 number over IAX?

Folks, with all due respect: this thread is now wy off topic, as it has nothing to do with Asterisk whatsoever. Please take it offline, or to ~biz. thx. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Re: [asterisk-users] How to park calls on a specific extension

Lacy Moore - Aspendora wrote: I think that's the problem with the Asterisk community right now. Anytime something is suggested, the response is either write it yourself or deal with what is there. Do you have experience with other big, complex Open Source projects? Do you know of any whe

[asterisk-users] Anyone got a dialplan for SPA ATAs for ISN?

After John Todd's talk at Astricon about the ISN project, I spent much of the weekend playing around with it. I have discovered that the default dialplans on my Sipura gear, as well as my Grandstream phones, intercept the "*" key that is a required part of ISN numbers and interpret it as a "me

Re: [asterisk-users] "No Authority Found"

Andrew Joakimsen wrote: Another thing is my understanding of the "peer," "user" and "friend." I thought that a "peer" can only receive calls from either a user or a friend, a "user" sends calls to a peer or friend and a friend is both a peer and a user, however in my production machine I hav

Re: [asterisk-users] Embedded Asterisk

Kristian Kielhofner wrote: Administrator TOOTAI wrote: Cory Andrews wrote: I caught a thread the other day concerning Astricon and users embedding Asterisk on a Linksys or Netgear broadband router. I lost track of the email thread, if anyone is presently working with this scenario please sh

Re: [asterisk-users] Netgear WGT Flash-fest at Astricon

Kristian Kielhofner wrote: Do you have the necessary components for a serial cable for these little guys? I would like to play with the loader and get a serial console... If you don't have one perhaps we can work on getting the parts before then. I have one, and also will be br

[asterisk-users] Netgear WGT Flash-fest at Astricon

Just an FYI to anyone out there who will be attending Astricon and who would like to play around with embedded Asterisk on the Netgear WGT634U platform. If you want to "bring your own" to the show, I'll be bringing all the appropriate stuff to flash them there with my latest openWGT/Asterisk

Re: [asterisk-users] Why is this happening?

Matt wrote: Ok I understand all that... Just wanted to confirm that A) it was the remote router mangeling the port and B) that it wouldn't cause an issue (I wasn't 100% sure if it would.. since only the 4569 port is open on the firewall). Could this cause an issue? If only 4569 is open on the f

Re: [asterisk-users] VoipSupply? [Semi-Urgent]

Jay R. Ashworth wrote: On Mon, Oct 16, 2006 at 10:21:29PM -0400, C F wrote: On 10/16/06, VaibhaV Sharma <[EMAIL PROTECTED]> wrote: I don't think this is a problem because of the snow storm. Yo, all; please take this thread where it belongs, which is the -biz list. It is not relevant to th

Re: [asterisk-users] Reception Console

Scott Higginbotham wrote: I'm interesting in testing this. OFF LIST PLEASE, FOLKS!! The list has enough traffic without the 10,000 "me too" mails that are likely to follow if nobody points out that it's bad netiquette. B. -- This message has been scanned for viruses and dangerous content

Re: [asterisk-users] How big is *your* dialplan??

Douglas Garstang wrote: I don't get it. The clients are ok with their phone systems being down anywhere from minutes to hours? Try googling for "cost benefit." I got 135 million hits. Your brain has some very odd twists in the way it works. Or you're a troll. B. _

Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list

Issac Simchayof wrote: Polycom 601 with Sip 2.01 Anyone using Sip 2.01? I have upgraded my phones and now presence no longer functions. Buddy list shows all phones online but status does not change when someone is on a call. Also blf does not function. I am using trixbox, 1.67 was working fine

Re: [asterisk-users] Re: Real-time and priority "n"

Tony Mountifield wrote: In article <[EMAIL PROTECTED]>, Ronald Wiplinger <[EMAIL PROTECTED]> wrote: Is it exclusive? Either Realtime or priority "n" ??? If so, what is the better way? I believe 'n' is just a shorthand way of writing "previous line + 1", and gets converted into an actual num

[asterisk-users] New ptlib dependency-requirement in SVN-trunk?

Suddenly I can't get SVN-trunk to build anymore; the configure script is looking for something related to ptlib I don't have: checking for /root/pwlib/include/ptlib.h... no checking for /usr/local/include/ptlib.h... yes checking for ptlib-config... no checking for ptlib-config... no Cannot find

Re: [asterisk-users] SER with multiple asterisk deployment

Douglas Garstang wrote: If your referring to using AVP operations to peek into the SIP message, and determine state, good luck finding documentation on that! From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Douglas Garstang wrote: > It won't work, unless you make sure that

Re: [asterisk-users] Re: Very high ping times from 7960 phones

Michiel van Baak wrote: On 09:42, Mon 25 Sep 06, Tomislav Par?ina wrote: In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... I'm sure other people are using 7960 phones so maybe someone could have a quick look at what time sip show peers reports? When I do a 'sip show peers' all my cis

Re: [asterisk-users] RE: 1.4 Beta 2 Config Problem

Andres wrote: Andres wrote: exten => 8600,13,GotoIf($[${CALLERID(number)} = 2013]?50:51) exten => 8600,50,Set(CALLERID(number)=2000) You are missing a parenthesis. It should read: exten => 8600,50,Set(CALLERID(number)=2000)) I just tried with a fairly recent version of SVN head--a clos

Re: [asterisk-users] (no subject)

[EMAIL PROTECTED] wrote: Hi, Looking for good rates for UK Landline & Mobile. Plus Saudi Arabia, UAE, India & Pakistan. This is a -biz question, not -users. Also, do you realize how bad it makes you look that you can't even bother to put a subject on your mail? B. -- This message has bee

Re: [asterisk-users] Digium GUI?

Douglas Garstang wrote: I wonder if the look and feel of this GUI will be completely configurable. If it's not, then I really don't think that's very useful. Service providers wouldn't be able to use it to let their customers manage their own settings, and customers wouldn't want to use it if

Re: [asterisk-users] Realtime Extensions -- Comments?

Douglas Garstang wrote: It doesn't matter where you turn in Asterisk, there's gotcha's. For example, you can't put the hint stuff into realtime, and there's no inherint way to comment extensions. It doesn't seem like Asterisk is good enough for you Doug. Switch to one of the competitors' p

Re: [asterisk-users] Realtime and labels

Douglas Garstang wrote: Does anyone know if realtime extensions support the use of labels? I don't believe so. As I understand it, the dialplan parser internally converts n-type and labeled priorities to a straight numeric format, which is then used internally. Becuase the Realtime engine

Re: [asterisk-users] Manager Interface API's

John Novack wrote: I, for one, didn't take his comment as anything other than constructive Lack of documentation is an issue, open source or not. It is an unfortunate situation that many very smart coders understand what they have created, but are unwilling or unable to supply enough informatio

Re: [asterisk-users] Manager Interface API's

Douglas Garstang wrote: Can anyone recommend the best Manager Interface API, putting language preferences aside? The python and perl ones have bupkiss documentation. I can't understand why anyone would even write an api and make it publically available without documenting it. Have you tak

Re: [asterisk-users] Ever donate Software to Digium? If you did your a fool.

EVERYONE PLEASE DON'T FEED THE TROLL!! That post was done only for the sake of generating responses, and we do no one any favors by taking the bait. thx. B. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To

Re: [asterisk-users] Garbled initial voicemail prompt

Joshua Colp wrote: - Original Message - From: Frank Tarczynski [mailto:[EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thu, 03 Aug 2006 07:16:15 -0300 Subject: [asterisk-users] Garbled initial voicemail prompt I'm having a problem where the very first words of the Asterisk

Re: [asterisk-users] RDNIS and IAX2

Douglas Garstang wrote: I'll probably get blasted for this. I hope I'm wrong, and then a little blasting is ok. It appears that Asterisk may have let us down again as a 'carrier grade' solution. Did the list software screw up, or did you post this exact same mail yesterday? B. __

[asterisk-users] Missing close quote in CallerID breaks SIP. . .workaround?

I posted about this some while back, and at that point was told "the remote end is broken, nothing we can do about it." The problem: for whatever reason, some CallerID names come in broken. There is an example CLI trace shown below. My question: is there anything I can do to fix this, since t

Re: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk

Douglas Garstang wrote: Here's my invite Brian. The From: is always going to contain the auth id the ATA used to register with Asterisk. INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK208dd2be;rport From: "Cody XXX-527-7107" ;tag=as3a94778b To: Conta

Re: [asterisk-users] Codec Negotiation

Douglas Garstang wrote: Would you like me to dig up the posts from Keving Fleming stating that this is known not to work Brian? As I recall those posts have to do with the way your particular setup required ARA to work with a failover/redundant cluster system you were building. Beyond t

Re: [asterisk-users] Codec Negotiation

Douglas Garstang wrote: Can't put it in a realtime database. We have multiple Asterisk boxes in a cluster, and it's a well known fact that multiple Asterisk boxes using realime cannot query a common MySQL database. Sounds crazy, but true. You spread some amazing "well-known facts" on this li

Re: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk

Douglas Garstang wrote: I'm working with a Sipura 3000 ATA here. I'm trying to get incoming PSTN calls on the FXO port to go automatically to Asterisk. I have it working, but I had to configure the ATA to register with Asterisk, which means that all calls are being sent to Asterisk with a call

Re: [asterisk-users] Re: asterisk-users Digest, Vol 24, Issue 116

Nobody is going to pay much attention to your help requests if you can't even figure out how to set it up so the subject header reflects what the problem is all about. Why don't you try again with an appropriate subject? B. -- This message has been scanned for viruses and dangerous content by

Re: [asterisk-users] Please suggest me Best VoIP Service Provider

Take this to the -biz list, PLEASE. B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRI

Re: [asterisk-users] Using dproxy to solve "no DNS hangs everything"problem?

C F wrote: Thanks Brian for your work, I have had the same problem I installed dnsmasq and I *think* the problem is gone now, I'm repeating I think, I'll only know when the internet goes down again. Your post inspired me to do one more little test: I removed the default route from my Asterisk

Re: [asterisk-users] Using dproxy to solve "no DNS hangs everything"problem?

James Harper wrote: This is a problem that affects more than just asterisk, so I'm sure there are solutions out there! One thing you might do is to put a trailing '.' on all fully qualified DNS names. Without the '.', the system will first try appending the default domain(s), which will cause

[asterisk-users] Using dproxy to solve "no DNS hangs everything" problem?

The subject pretty much says it. Wondering if there's anyone out there who, as an alternative to hard-coding IP addresses in /etc/hosts, has implemented dproxy or somesuch to enable Asterisk to survive DNS outages. I had a royal mess on my hands this morning after my Internet connection went

Re: [Asterisk-Users] app_conference DTMFs?

Henry J. Cobb wrote: I tried several different combinations of app_conference and Asterisk versions and then I had to get back to actually providing phone service that didn't crash. I hate to me-too, but my experience was identical. Crash after crash, and I tried everything that was suggest

Re: [asterisk-users] Re: for you guys setting up customer offices...

Benny Amorsen wrote: "BC" == Brian Capouch <[EMAIL PROTECTED]> writes: BC> Everyone's mileage varies, and IMO it doesn't do any of us any BC> good for negative opinions to be presented to the public as fact. BC> You disclaimed, indeed, but you would have b

Re: [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls to Asterisk

Thomas Kenyon wrote: For some reason when I do this, It only works if I have callerID switched off, otherwise I get authentication errors. Do you know of anyway to bulk-save the contents of all the config screens on that unit? If so, I could scrub the passwords and send you the config for

Re: [asterisk-users] OT: Sipura SPA-3000 ATA Directing Calls to Asterisk

Douglas Garstang wrote: Somewhat off topic... I have a Sipura SPA-3000 ATA. Calls are coming in from the FXO port. I'm trying to get all calls forwarded to Asterisk. However, (and this is hard to believe), the docs say that 1-stage calling (I presume that means no PIN is required) is not poss

Re: [asterisk-users] for you guys setting up customer offices...

calvis wrote: Grandstreams are junk. (I have had bad experiences with them) The former doesn't necessarily derive from the latter :-) Others of us have found them to be an excellent low-cost solution that puts VoIP in places it otherwise would not be economical to deploy. Everyone's mile

Re: [Asterisk-Users] Best VoIP provider for Asterisk

[EMAIL PROTECTED] wrote: Termilink, at www.termilink.net Just out of curiosity, would you happen to be affiliated with this provider? If you get my drift. . . . B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed

Re: [Asterisk-Users] Re: Digium Hardware Reliability

M.Hockings wrote: Mike (totally UNimpressed with Digium) Point taken. I was not so much point fingers but asking what my expectation should be and maybe shedding some frustration. I don't really have a lot of experience with this kind of communications gear All the more reason for

Re: SV: [Asterisk-Users] Error in config sample for GoToIf?

Jon Schøpzinsky wrote: Hello As far as ive understood, you can just write Exten => s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail) ${AVAILSTATUS} would return 1, and "${AVAILSTATUS}" would return "1" Through more testing, the double quotes I used seemed superfluous; if you use them in bot

Re: [Asterisk-Users] Error in config sample for GoToIf?

Brian Capouch wrote: exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail) I couldn't get this to work unless I surrounded the first part of the test with quotes, too, like this: exten => s,n,GoToIf(["${AVAILSTATUS}" = "1"]?autoanswer:fail)

[Asterisk-Users] Error in config sample for GoToIf?

My teeth are on edge after this one. A couple of perfectly good hours of my life, and I still don't know what's going on. . . . The extensions.conf.sample that comes with the current SVN trunk has this line, in an example that shows how to use ChanIsAvail: exten => s,n,GoToIf([${AVAILSTATUS}

Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!

Francesco Peeters (Asterisk) wrote: Ja dat kun je wel zeggen ja... Maar goed dat Nederlanders vrij aardig Engels praten! ;-) Pues my punto fue que un poquito de correo en otro idioma no hace daño, y si ayuda mucho y molesta poco, ¿por qué quejarse? B. -- This message has been scanned f

Re: [Asterisk-Users] AsteriskPT- Sucessfull routing Skype Calls to my * box. Incoming and Outgoing calls!!!

Tzafrir Cohen wrote: On Mon, Jun 26, 2006 at 09:39:11AM -0300, Josué Conti wrote: Marco, bom dia. Essa interligação entre o Skype e Asterisk, é feito atavés de um módulo externo? É freeware? Podemos seguir com o projeto Asterisk-PT? English, please, folks. Let them talk. What's it hurt t

[Asterisk-Users] Oh oh. Micro$oft just noticed VoIP

It will be interesting to see how many standards get broken, and how many proprietary hooks get thrown into the pot. The bean counters smell some money, and their OS franchise is waning: http://www.nytimes.com/2006/06/26/technology/26soft.html -- This message has been scanned for viruses and

Re: [Asterisk-Users] Upgrading old version of Asteriak - changes

Martin Joseph wrote: Huh, I never looked at that file before (phone.conf). Actually they seem to refer to a Linux telephony interface? Anyone please care to elaborate on what the phone.conf file is really for? The wiki just has a copy of the default file... It's the conf file for the

Re: [Asterisk-Users] Don't use CDRTool From AG-projescts

[EMAIL PROTECTED] wrote: hello to all, I advice you to not use Harry!! Only one post is needed for each of your silly complaints. Please don't give people even more reason to relegate you to their killfiles. B. -- This message has been scanned for viruses and dangerous content by MailSc

Re: [Asterisk-Users] syntax error

Rob Thomas wrote: And once again I am reminded why I shouldn't bother helping people here. Not even a 'thanks'. Dang, dude. If everyone who got helped on this list responded to the list with a thank-you, list traffic would go up by at least 10% :-) And it's a PIA to manually circumvent t

Re: [Asterisk-Users] Anyone using VoIP WiFi phones?

Warren wrote: As this is to be used by company execs when travelling that is not a viable option. The Netgear WIP300-NA looks interesting. Amyone used that one yet? Sorry this is on-topic for the thread but not what you've written up there. I have some Starcom F1000s (802.11b model). They a

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