Re: [asterisk-users] Call abandoned from queue not showing in CDR (possible bug)

2012/1/12 Ishfaq Malik > Hi > > I'm using 1.8.7.0 with the RealTime architecture. > > If a call goes into application Queue and is abandoned by the caller, no > entry is made in the CDR. Entries are made into the queue log. > > This cannot be correct behaviour, all calls should show in the CDR. >

[asterisk-users] "intercom" SIP header being ignored by Kirk wireless handsets

. We are running Asterisk 1.4.23.1 (TrixBox CE). We are running latest stable firmware on the handsets. Most other features on handsets seem to work fine except this. We followed the instructions in the deployment doc' which had us send the "intercom" keyword in the SIP header, as seen

Re: [asterisk-users] fraud advice

managed > to guess one of the SIP passwords. 4000 calls to various middle eastern > destinations have been placed, which ended up being sent over our > customer's PSTN trunk, and of course there was no warning until the bill > came today. Unfortunately the bill only covered the fir

Re: [asterisk-users] billsec exceeds duration on some calls

On 8/20/10 1:24 PM, A J Stiles wrote: > On Wednesday 11 Aug 2010, Tilghman Lesher wrote: >> On Wednesday 11 August 2010 03:59:28 A J Stiles wrote: >>> I'm having a problem with Asterisk 1.6.2.9 with the MySQL cdr addon. >>> >>> With some calls, the value in the `billsec` field in the CDR is excee

Re: [asterisk-users] spam blacklist

Supposedly, the filters drop it in the transaction stage. But for some reason, every time I get dropped from the list, it's just after a spam email was sent out en masse, so I'm not sure what's up there. On 7/28/10 10:43 PM, jon pounder wrote: > SIP wrote: > what can yo

Re: [asterisk-users] spam blacklist

On 7/28/10 9:45 PM, Sam wrote: > Just a note, the asterisk mailing list server continually gets > blacklisted over at > http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering > mail to spamtraps. Perhaps something needs to be looked into... > > Regards, > Sam > Spammers sign up

Re: [asterisk-users] Need USA DIDs

On 6/23/10 7:20 AM, RSCL Mumbai wrote: > Hi, > > Looking for some reliable and quality providers of USA DIDs. > > Any pointers ? > > Thx > Sans We've had good luck with Vitelity and DIDForSale.com. N. -- _ -- Bandwidth and Colo

Re: [asterisk-users] Small PC to build and run Asterisk

Danny Nicholas wrote: > Also cheaper to replace flash card than hard drive. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet > Sent: Monday, June 14, 2010 4:21 PM > To: asterisk-users@lists.d

Re: [asterisk-users] Small PC to build and run Asterisk

w power consumption > * No fan or very small fan > * Hard drive (not flash memory) > > Capabilities/capacity > > * No GUI, no X > * Register to multiple SIP servers > * There will be no PSTN > * No analog phones > * Small number of SIP devic

Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ?

Jeff LaCoursiere wrote: > On Thu, 20 May 2010, Gordon Henderson wrote: > > >> On Thu, 20 May 2010, SIP wrote: >> >> >>> Even IF you could get a keyboard with lights you could individually turn >>> on and off (never seen one), >>>

Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ?

Tzafrir Cohen wrote: > On Wed, May 19, 2010 at 07:46:26AM +0200, Olivier wrote: > >> 2010/5/18 Danny Nicholas >> >> >>> Dumb question – wouldn’t it be easier to monitor a web interface than a >>> phone with 100 lights? >>> >>> >> Yes and no : operator already has a Flash Operator P

Re: [asterisk-users] AGI <==> DeadAGI

On 5/2/2010 4:52 PM, Steve Edwards wrote: > On Sat, 1 May 2010, SIP wrote: > > [snip] > > >> We run DeadAGI for a considerable number of calls since it has the >> ability to run post-hangup cleanup no matter which side hangs up (unlike >> AGI). >>

Re: [asterisk-users] AGI <==> DeadAGI

On 4/30/2010 6:03 PM, Luki wrote: >> It is irrelevant who hangs up, you want to just use DeadAGI in the h >> extension >> > I wish that would be the case, but at least on 1.4 I see: > > [Apr 30 14:59:38] -- Executing [...@master-route:1] DeadAGI(...) in new > stack > [Apr 30 14:59:38] WA

Re: [asterisk-users] Wanted: free DID number and provider feedback

> My bad, I'm in Los angeles california usa > > On Thu, Mar 18, 2010 at 1:06 AM, SIP wrote: > >> What country are you in? Makes somewhat of a difference. >> >> N. >> >> >> On 3/17/2010 8:49 PM, Mike wrote: >> >>> Ok, I see there

Re: [asterisk-users] Wanted: free DID number and provider feedback

What country are you in? Makes somewhat of a difference. N. On 3/17/2010 8:49 PM, Mike wrote: > Ok, I see there's alot out there of voip providers. > > Curious what to watch out for ? charges and fee's, etc ? > > If anyone has feedback as to a GOOD voip provider experience (one that > gave FREE

Re: [asterisk-users] MWI and 1.6.1

Will Payne wrote: > it just seemed like a 'I know this is wrong, but...' comment :) > Quoting entire emails is bad, m'kay. Quoting whole threads is worse. If you > snip the quote down to the relevant portion, you can reply where you like, > regardless of what's gone on beforehand. > > (Surely th

Re: [asterisk-users] MWI and 1.6.1

Will Payne wrote: > On 8 Mar 2010, at 22:08, Dave Poirier wrote: > > >> Top posting to remain consistent... >> > > > I drop litter because everyone else does. > > ;) > > W > > Different entirely. People who switch to bottom posting on a top-posted thread make things MUCH harder to read

Re: [asterisk-users] Ideasip

David @ULC wrote: > > I use IdeaSip with IPKall. > > How may channels are open when we use IdeaSip ? > Incoming IdeaSIP SIP channels are unlimited; however, I believe IPKall limits you to 94 channels via their DIDs. You would, of course, need the bandwidth to be able to handle 9

Re: [asterisk-users] GXV3140 and Xlite video

Julian Lyndon-Smith wrote: > Has anyone managed to get these two phones to make a video call to each other > ? > > If so, care to share how the hell you managed ? > > the GXV is at the latest firmware, and xlite the latest download > > Asterisk 1.4 trunk > > TIA > > Julian > > Yes. Have done it

Re: [asterisk-users] Please some enlightment on ENUM !!

John Novack wrote: > Raimund Sacherer wrote: > >>> Adding random digits to a PSTN and expecting to get the same person at a >>> different extension you don't think that's a hack? I do. One should >>> >>> >> Sorry, please do not call a whole country using a hack when their soluti

Re: [asterisk-users] Please some enlightment on ENUM !!

Raimund Sacherer wrote: >> Adding random digits to a PSTN and expecting to get the same person at a >> different extension you don't think that's a hack? I do. One should >> > > Sorry, please do not call a whole country using a hack when their solution is > legitimate. > > Austrian PSTN >

Re: [asterisk-users] Please some enlightment on ENUM !!

Benny Amorsen wrote: > SIP writes: > > >> It may work in Austria, and may even be valid in Austria. But if that's >> the case, it's because Austrian dialing is a complete hack -- NOT >> because that's the way it's intended OR designed. >

Re: [asterisk-users] Please some enlightment on ENUM !!

Philipp Kempgen wrote: > Leif Neland schrieb: > >> Norbert Zawodsky skrev: >> > > >>> The number +43-1-3207978 is my telephone number. I "own" it as long as I >>> pay for it. And with extra digits behind it I can do whatever I like. I >>> can create any extension - physical or virtual. I

Re: [asterisk-users] Please some enlightment on ENUM !!

e world can call > me using this number. > Lets say, I have 3 extensions: 0=reception, 10=secretary, 20=boss. > > If someone calls > > ENUMLOOKUP(+4311234567) he will get a uri "sip:0...@ip.of.my.asterisk" > ENUMLOOKUP(+43112345670) he will get a uri "sip:0...@ip.of.m

Re: [asterisk-users] Please some enlightment on ENUM !!

quot; both return NAPTR records. >> >> Now for the less clearer points: >> >> Your'e supposed to register your number without any extension. >> If I have some extensions here, how can the calling party get the >> correct sip uri to the requested e

Re: [asterisk-users] solution for NAT issues?

Does the phone have some sort of NAT Keepalive setting? Often, the only way to keep that port open on the user's NAT gateway is to have the NATted client send the occasional data out through the port. N. Ron wrote: > i have also tried setting qualify='yes' but cpu usage spiked to 100%. > > Ron w

Re: [asterisk-users] allowguest defaults to yes for SIP

to. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Howard > Sent: Thursday, November 12, 2009 12:48 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re:

Re: [asterisk-users] SNOM 870

Remco Barendse wrote: > On Fri, 30 Oct 2009, hbk wrote: > > >> Hi, >> >> I have played with the 820 for some weeks, mostly love it excellent speech >> quality. Even got the "mini" browser running >> showing my favorite webcam, this could be put to real use too:) >> >> Issues so far: >> Some emb

Re: [asterisk-users] Astricon

Sounds like it wasn't a very interesting track. ;) N. Danny Nicholas wrote: > Is THAT a summary :)? > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Randy R > Sent: Wednesday, October 21, 2009 1:24 PM >

Re: [asterisk-users] Calls hang up after 20 seconds

Alex Balashov wrote: > SIP wrote: > > > > What is your citation for this qualification? RFC 3261 does not seem to > me to say that, as in 13.1: > > Because of the protracted amount of time it can take to receive final > responses to INVITE, the reliab

Re: [asterisk-users] Calls hang up after 20 seconds

Kevin P. Fleming wrote: > SIP wrote: > > >> In an ideal world, when Asterisk sent an ACK, whatever server/client it >> was connected to would respond accordingly. It is, however, not an ideal >> world, so this doesn't always happen. >> > > This i

Re: [asterisk-users] Calls hang up after 20 seconds

Gianni Fioretta wrote: > Hello. > > I have a problem with Asterisk, sometimes it hangs up an external call after > 20 seconds, apparently without any reason. > The call comes from a SIP server hosted from EuteliaVoIP, many peers rangs > and one of them answer, the call en

Re: [asterisk-users] Best Firewall Suggestions?

Better throughput than a pix, worlds easier to operate and configure, and comparable in price. Very SIP/VoIP friendly. Loads of optional modules (we use its mail filter module to filter spam/viruses for several hundred thousand user mailboxes, for instance) to limit the cost to what you need. Also has

Re: [asterisk-users] VUC: RE: Friday 11th: Aswath Rao: "Trapezoidal VoIP is Evil" on VoIP Users Conference at Noon EDT

n.net >> +1-212-203-4357 New York >> +61-2-9016-5642 (Sydney in-dial). >> +44-20-3129-6001 (London in-dial). >> >> > > I don't know if I'll be able to make the call but my guess is he's > referring to the SIP trapezoid: > > http

Re: [asterisk-users] Allowing multiple callers to join a public speaking session...?

An Asterisk MeetMe conference sounds like the ideal sort of scenario for you, allowing people to join in or drop off during a session as they please. N. li...@mgreg.com wrote: > Hi All, > > As is obvious by my joining the list, I'm interested in learning more > about Asterisk. I have downlo

Re: [asterisk-users] How to deal with PayPal frauds?

When you start taking credit card payments (assuming you will), be careful with small payment amounts. You'll become a fraud haven. A lot of CC thieves or people who've just bought a CC number will use a small amount charge to check and see if the card is any good. Check out some of the MaxMind st

Re: [asterisk-users] IPKall and FWD

Means your username is not registered on the IdeaSIP system (your client/phone is not logged into IdeaSIP). N. David @ULC wrote: > " you're not logged in " means ? > > > On Mon, Aug 24, 2009 at 11:39 PM, David @ULC > wrote: > > > > Oh my god.. > > Today its s

Re: [asterisk-users] IPKall and FWD

A quick look at the system shows you're not logged in, which is why you're getting that message. N. David @ULC wrote: > > > Oh my god.. > > Today its saying there is NOONE to take your call.I am using IdeaSIP > > What could be the reasons ? > > It was working perfectly till saturday . > > > On T

Re: [asterisk-users] IPKall and FWD

IdeaSIP, GizmoProject, IPTel, maybe OnSIP (don't quote me on that one, I'm not sure, but someone around has surely used it), etc, etc. There are a lot of alternatives about. Disclaimer: IdeaSIP is my personal unruly child (hence top billing on the list of alternatives). N. David @ULC wrote: >

Re: [asterisk-users] Accessing to ekiga.net through Asterisk

gt; -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > SIP wrote: > > >> Daniel, >> > > Hi SIP. > > >> Check your stunaddr setting. Is it misspelled, or do they really use >> stun.exiga.net instead of stun.ekiga.net ? >>

Re: [asterisk-users] Platform decision ...

ther. And even when you do, it will function screamingly > well. > But it won't have doors, windows, AC, or creature comforts that we've > all come to expect. > > > You mean comforts which you have come to expect. Again, my needs have > been so far fulfilled

Re: [asterisk-users] Platform decision ...

"http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ";): > > The main points listed on Asterisk's "CONS" that concerned me were: > > * Conferencing on Asterisk depends on Zaptel hardware and/or kernel > modules for timing; > * Lack of buil

Re: [asterisk-users] Accessing to ekiga.net through Asterisk

n.exiga.net > insecure=port,invite ; required for incoming ekiga.net calls > > /etc/asterisk/extensions.conf: > > [from-internal] > ... > exten => _8.,1,Dial(SIP/ekiga/${EXTEN:1},20,r)) > > > I tried a echo test, dialing in my case to 8500, but in spite of seeing > t

Re: [asterisk-users] Time of Day Routing

Tony Mountifield wrote: > In article <05d03313-994b-4892-b045-f61332ddb...@geekinter.net>, > Steve Howes wrote: > >> On 14 Aug 2009, at 09:17, Neeraj Chand wrote: >> >> >>> Asterisk version 1.4 >>> From: Neeraj Chand >>> Sent: Friday, 14 August 2009 8:17 PM >>> To: 'asterisk-users@lists.di

Re: [asterisk-users] Different codecs for reading and writing

I'm not sure there IS an issue, per se. There are lower bitrate codecs that will work fine for voice communications in both directions. But if you're trying to force a low-end codec to the upstream, that just means the downstream on the remote end is going to be stuck with a low-end codec. And if h

Re: [asterisk-users] Asterisk on OpenWRT

I've had similar results to you. Packet loss even when not transcoding. Overall poor performance across the board. We considered it a failed experiment. N. Zoa wrote: > I have played with DD-WRT on linksys wrt54g version 5 last week (2 > different ones, they are the model with less memory so

Re: [asterisk-users] best practices for running asterisk as SIP B2BUA

Alex Balashov wrote: > BTW, if you need a generic, scalable, high-volume B2BUA, it is not a > "best practice" to use Asterisk for that purpose. > > Indeed. But you can grow some good SMB B2BUA systems out of it. Freeswitch would be a grand alternative... if it had documentation. Anywhere. Ever

Re: [asterisk-users] [Fwd: confirm f1ab6c493110edited]

Dunc wrote: > Doug Lytle wrote: > >> >>Your membership in the mailing list asterisk-users has been disabled >> >> due to excessive bounces The last bounce received from you was dated >> Anybody else seeing this? My mail server logs don't show any issues. >> >> Doug >> >

Re: [asterisk-users] SIP and FW settings

Michael wrote: > On Tue, 14 Apr 2009 20:47:29 you wrote: > >> Hi michael, >> >> you should open both tcp,udp 5060,5061 too and as you mentioned between >> 1-2. >> > > AFAIK 5061 TCP is for TLS SIP which isn't used much yet? > >

Re: [asterisk-users] IPkall

Daniel Nowacki wrote: > SIP wrote: > >> IPKall still exists. >> >> http://www.ipkall.com >> >> No customer service, and the number has to be used every month or you >> lose it. But it's there. And free. And good. >> > > I get an ug

Re: [asterisk-users] IPkall

IPKall still exists. http://www.ipkall.com No customer service, and the number has to be used every month or you lose it. But it's there. And free. And good. N. Dean Collins wrote: > > Does IPKALL still exist? > > I am after a free SIP trunk – who is still giving these away t

Re: [asterisk-users] Asterisk Security

nce we are talking about security, if I am using * to talk to a cisco > gateway via SIP, is there some sort of encryption you can use? Like a > vpn tunnel? > > Can someone capture packets and re-assemble to make out a conversation? > > > > -Original Mes

Re: [asterisk-users] Is there a public blacklist of hackers' IPaddresses?

Dave Platt wrote: >> SIP was written in such a way that the hashes it sends for passwords >> could, with only a trivial rewrite of the server code, be SHA1 instead >> of MD5 -- which would increase security to the level that, currently, it >> would be far more trouble

Re: [asterisk-users] Is there a public blacklist of hackers' IPaddresses?

randulo wrote: > On Thu, Mar 26, 2009 at 4:19 PM, SIP wrote: > >> The first approach is the current approach: build software with little >> thought to how it will be secured, opting for all the work of securing >> > > What about SIP itself? Does it provide

Re: [asterisk-users] Is there a public blacklist of hackers' IPaddresses?

randulo wrote: > On Thu, Mar 26, 2009 at 2:38 PM, SIP wrote: > >> And so, in answer to your question, I don't think there ARE necessarily >> steps that can be taken right now to ensure that there's a rational >> approach to the resolution of such an issue

Re: [asterisk-users] Is there a public blacklist of hackers' IPaddresses?

randulo wrote: > On Thu, Mar 26, 2009 at 1:32 PM, SIP wrote: > >> As an end-point ITSP, I can assure you, it would be us who's assessed >> the requisite charges. If someone uses a fraudulent card, we're required >> to pay. If someone uses a three letter

Re: [asterisk-users] Is there a public blacklist of hackers' IPaddresses?

randulo wrote: > This brings up a side issue. Banks on the Internet have had to provide > a sort of insurance that allows the customer to be protected if > someone hacks in to his or her account. ITSP will need to think > carefully about having a similar policy that protects people from an > attack

Re: [asterisk-users] Good phone near $125

David Ruggles wrote: > I was looking at the aastra 9133i, however I was informed that this phone is > no longer supported. What are good phones around the $100 - $125 price > point? (Need POE) > > Thanks, > > David Ruggles > CCNA MCSE (NT) CNA A+ > Network Engineer Safe Data, Inc. > (910) 285-

Re: [asterisk-users] IAX based war dialer

Not to burst your bubble, Jon, as I agree with a majority of what you said... but using an argument about the evolution of email to support an argument about how telcos should have better tracking and accountability is somewhat weird. We get 3 million email messages a day through our servers. 9

Re: [asterisk-users] building a phone

phone ergonomics, psychology, and reliance, and replaces it with something that's clearly just a kludgy add-on to a product which was never originally designed for the task. > One thing that bothers me with the current crop of hardware SIP phones > is that they are hopelessly properitary.

Re: [asterisk-users] AGI pdf book

Michael wrote: >> This has absolutely nothing to do with the fact that something is >> opensource. The fact that the source is "open" has nothing todo with its >> pricetag. Sometimes opensource products are more expensive then closed >> source products. >> >> If you want support/maintenance/dedicat

Re: [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed

Grygoriy Dobrovolskyy wrote: > > > 2009/2/13 Tzafrir Cohen > > > On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote: > > > > I've been involved with getting better data for running Asterisk on > > the Amazon EC2 cloud computing system. Here

Re: [asterisk-users] Amazon Flexible Payment System - micropayments finally cracked?

James Moore wrote: > Notice that one of the prohibited items is: > > # Phone Services - includes 800 or 900 phone services and audio text > services, prepaid phone cards, and prepaid phone services. > > https://payments.amazon.com/sdui/sdui/about?acceptableuse > > Google Checkout started with th

Re: [asterisk-users] RFC -- Improving the quality of the mailing lists

Ira wrote: > At 09:30 AM 1/27/2009, you wrote: > >> People are always going to ask stupid questions. >> > > For me it's not so much the stupid questions as the expectations that > we're here to solve their problems according to their needs. If that > continues to happen and the noise leve

Re: [asterisk-users] Root Password not taking

Steve Edwards wrote: > On Thu, 22 Jan 2009, Wilton Helm wrote: > > >> If some of your directories like /home and /user have separate mount >> points, they don't have to get wiped out in the process. >> > > If there is any reason to suspect a hack, re-installation is the only way. > I woul

Re: [asterisk-users] evaluate SIP response codes in dialplan

Take a look (if it still exists) at the Asterisk B2BUA project. It has a patch that adds direct access to SIP response codes. It takes a little modification of the patch file to use in some of the newer asterisks (and to strip out the one codec option that's somewhat irrelevant), but i

Re: [asterisk-users] What are the various models of DID providers

Alex Balashov wrote: > SIP wrote: > > >> What's interesting is the number of caveats and mixes even in the CLEC >> and ILEC world. I work with a CLEC that is also an ILEC (in certain >> areas), since they encompass various areas in Georgia (and own the >&

Re: [asterisk-users] What are the various models of DID providers

aching smaller > users increasingly for direct VoIP trunking. And of course, customers > with very large volumes of traffic can go to the carrier directly and > often do, if the business case for it is right. > > The VoIP wholesale DID providers traditionally interfaced with the &g

Re: [asterisk-users] Bring India together

Look, ma... spam! We dun never seen that 'n before. N. Sunkara RaviPrakash wrote: > > Hi, > > Imagine a billion Indians together. > > Already 3 million Indians have chosen Indyarocks.com to bring India > together. > > I am already part of it and dont be surprised if you find most of your > oth

Re: [asterisk-users] Dial timeout with SIP - how to set timeout for INVITE ACK

It's a valid concern, but be prepared for people to tell you that this should be done with the qualify parameter to determine if a host is up and running. Not the most ideal way to handle it, I'll agree. But the SIP proxy functionality of Asterisk is limited (as it's not intended to

Re: [asterisk-users] Dial timeout with SIP - how to set timeout for INVITE ACK

> > > *From:* asterisk-users-boun...@lists.digium.com on behalf of Philipp > Kempgen > *Sent:* Thu 18/12/2008 4:17 PM > *To:* Asterisk Users > *Subject:* Re: [asterisk-users] Dial timeout with SIP - how

Re: [asterisk-users] top posting again [was: Re: CDR Design]

Steve Edwards wrote: > On Wed, 17 Dec 2008, Danny Nicholas wrote: > > >> OSUR GONNA BE ABLE TO MAKE PEOPLE STOP POSTING. IF DIGIUM GETS ENOUGH OF >> THESE STUPID HITS, THEY WILL CUT THIS OFF. I KNOW I'M SHOUTING, I'M >> @#$###$# TIRED OF INTERRUPTING IMPORTANT WORK TO READ NOTHING. THAT'S WHA

Re: [asterisk-users] Country numbering plan resources

Jeff LaCoursiere wrote: > On Sun, 14 Dec 2008, Tzafrir Cohen wrote: > > >> Right. So for those of us who want to do simple things and avoid >> complicated stuff such as telephony in shoddy continent of North >> America, could you please provide data for your country? >> >> So far we have AU, IL

Re: [asterisk-users] Country numbering plan resources

Michael wrote: > > Yes, but with an A-Z carrier, this can become risky when landline calls are > charged very differently to cellular calls, as is the case in NZ, Australia > and many other countries, unless someone is just a 'virtual' provider and > letting their up line do the invoices. > >

Re: [asterisk-users] Using DECT phones as SIP phones?

Fred Posner wrote: > > On Dec 5, 2008, at 11:31 AM, Michael Graves wrote: > >> On Fri, 05 Dec 2008 10:59:54 -0500, Neil Fusillo wrote: >> >>> Michael, >>> >>> Was there something particularly special you had to do to get your M3 to >>> work? I'm now on my second one from E4 Technologies (from whom

Re: [asterisk-users] Friday, Asterisk is 9 years old!

; You can get all the dial in information at > http://VoipUsersConference.org including info on a SipAddHeader() > kludge to avoid DTMF problems. > > IRC is Freenode.net #voip-users-conference join this even if you > can't call in. > > Call via SIP: [EMAIL PROTECTED] (thanks to

Re: [asterisk-users] OT: What do you guys think of this?

Doug wrote: > At 07:00 12/2/2008, SIP wrote: > >Doug wrote: > >> At 18:56 12/1/2008, Tilghman Lesher wrote: > >> >On Monday 01 December 2008 06:21:33 pm Doug wrote: > >> >> We tell our customers that they are not allowed to > >> >

Re: [asterisk-users] OT: What do you guys think of this?

Doug wrote: > At 04:03 12/2/2008, Benny Amorsen wrote: > >Doug <[EMAIL PROTECTED]> writes: > > > >> "Net Neutrality" is great in principle. But ISP's need to > >> somehow control those few percentage of users who suck down > >> a huge majority of the bandwidth. It's dollars and cents. > > >

Re: [asterisk-users] OT: What do you guys think of this?

Doug wrote: > At 18:56 12/1/2008, Tilghman Lesher wrote: > >On Monday 01 December 2008 06:21:33 pm Doug wrote: > >> We tell our customers that they are not allowed to > >> download copyrighted material. > > > >So your customers are only allowed to download public domain > >material? That kin

Re: [asterisk-users] Load balancing Asterisk.

to know, F5 is a unix-based box, just like the others. Last we used the F5s, they were all running a slightly modified BSDI. And only slightly modified in packaging. As for the current F5 SIP load balancer, we tried it a few years back and it was a dismal failure. It wanted to do cookie-ba

Re: [asterisk-users] Load balancing Asterisk.

Unless the LB is SIP-aware, and can maintain a SIP session, I don't see how it would work. As the SIP command stream sends discrete commands, without some sort of basic level of session awareness, there's no guarantee over a reasonable-length call that the INVITE and BYE would even get s

Re: [asterisk-users] Any other "free" toll free SIP providers out there?

Tom Browning wrote: > > FWD (Free World Dialup) allows any SIP call to US toll free numbers > via [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> > This works WITHOUT the need to be registered at FWD so in my dialplan > I have something like: > > exten => _8.,1,Dial

Re: [asterisk-users] Outgoing SIP calls dropped after 30 seconds.

We are only using SIP > (3 trunks now, instead of 2) and having all 3 in use is not an issue. > > Problem: Make a call on a Polycom 320 IP phone to any number and (4/5 > times) it will drop the call after 30 seconds. I noticed that the > little timer that pops up on the LCD on the pho

Re: [asterisk-users] Spam from DIDForSale <[EMAIL PROTECTED]>

Greg Woods wrote: > On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote: > >> Gotta love this list being farmed for spammers now. I am sure they call >> it targeted delivery or some such nonsense. I can't wait for capitalism >> to completely fail, then there won't be any spam. >> > >

Re: [asterisk-users] VoIP Users Conference Call Friday Nov 7 On Wideband Voice & Conferencing

randulo wrote: > On Tue, Nov 4, 2008 at 5:00 PM, Michael Graves <[EMAIL PROTECTED]> wrote: > >> In any case, the wideband bridge for this weeks VUC call supports only >> G.722. >> > > But we do plan to make a recording of both conference version available, > AFAIK? > > r > > But will i

Re: [asterisk-users] Strange ring tone: Long-Short-Short

Joseph wrote: > I'm using Linksys SPA3102 adapter and have a strange ring tone: > Long-Short-Short or Long-Long-Short-Short > > Does anybody know which setting adjust this ring tone on SPA3102 > Sipura rings normally. I'm not sure if setting are on Regional Tab or User Tab > > Interestingly, I g

Re: [asterisk-users] How Secure Is Asterisk

It's not 100% secure. Like any dual-key encryption, it's subject to the classic man-in-the-middle attack. This is why the Windows Zfone app has the addition of a visual key you can read and coordinate with the recipient to determine if a MITM attack is occurring. But only if you know what you're do

Re: [asterisk-users] SER + Asterisk

Alex Balashov wrote: > SIP wrote: > > >> Seriously, though... this seems to be a popular misconception. I hear it >> a lot. Where did you come across the information that SER is no longer >> developed? >> > > That seems to be a consequence of looki

Re: [asterisk-users] SER + Asterisk

etting the correct meaning of 'defunct', but from the >> last part of your suggestion i guess you value Kamailio/OpenSIPS more >> than SER. >> >> Are there some hard reasion for this. >> >> I am in the process of deciding which SIP server i want to use

Re: [asterisk-users] No reply to our critical packet

k. I suppose I could code around it, but not being the world's most proficient C coder, I'm always afraid I'll break something else. ;) N. Andrew Joakimsen wrote: > I am using a Polycom 501 SIP phone behind NAT. Asterisk server is > public with no NAT... everything works o

Re: [asterisk-users] OT: text/plain

Philipp Kempgen wrote: > Andrew Kohlsmith (lists) schrieb: > >> On October 5, 2008 12:22:37 pm Philipp Kempgen wrote: >> > > >>> ---cut--- >>> http://lists.digium.com/pipermail/asterisk-users/2008-October/219538.html >>> http://lists.digium.com/pipermail/asterisk-users/2008-October/21954

Re: [asterisk-users] OT - Is sip.instance useful ?

simple registration parameters using IP/port combinations as differentiators, but if you're running a symmetric NAT, that may be misleading or even non-functional. The instance COULD act as an additional identifier to help clarify those situations on the SIP side (as opposed to just the RTP side).

Re: [asterisk-users] Knowing incoming call technology and channel [SOLVED]

Eric "ManxPower" Wieling wrote: > Olivier wrote: > > >> I don't have any spare zaptel enabled system I could try this on, but I >> was not aware of this CHANNEL variable. >> Now, I can see it here http://www.voip-info.org/wiki/view/Asterisk+variables >> Maybe, I will add a line in www.voip-info

Re: [asterisk-users] sip forking needed for ekiga 3.0

Brian J. Murrell wrote: > On Fri, 2008-09-26 at 10:16 -0400, SIP wrote: > >> The RFCs are there for a reason. All SIP forking is UAS territory. Not >> UAC territory. >> > > From http://bugzilla.gnome.org/show_bug.cgi?id=553810 Damien Sandras > asks: >

Re: [asterisk-users] sip forking needed for ekiga 3.0

x27;) matches the default gateway, your machine is misconfigured and internet traffic will not properly flow. I know you're just the messenger here, and it's not your fault. But the message is wrong. Ekiga has tried to solve a problem (that of determining a 'best path' for SIP to

Re: [asterisk-users] sip forking needed for ekiga 3.0

Alex Balashov wrote: > You need to define what you mean by "SIP forking." There are many > things people mean by that. They are usually one of: > > 1) Call branching (proxies do this). > > 2) Parallel but distinct call legs managed by a UAC (this is what >

Re: [asterisk-users] sip forking needed for ekiga 3.0

Here's a bunch of packets. Figure out what to do with them. I'll be waiting for your response." There's a reason routing rules exist and mature services allow you to control the interface from which it originates. N. Brian J. Murrell wrote: > On Thu, 2008-09-25 at 1

Re: [asterisk-users] sip forking needed for ekiga 3.0

My thoughts are that to do parallel requests from every IP address on the machine is extremely weird behaviour. How would any server know which to respond to? SIP forking is supposed to send requests to multiple different destinations (or fork mid-stream to send to different destinations

Re: [asterisk-users] What is in practice the maximum no of simultaneous calls that Asterisk 1.4 can handle

It's common sense. Using all iLBC, I can't seem to get 100 simutaneous calls on my AMD 486 dx2/66. I don't get it! ;) N. Eric "ManxPower" Wieling wrote: > Where did you hear this? > > Shaun Wingrin wrote: > >> I have heard it said that, Asterisk falls over at 100 simultaneous >> calls. Is

Re: [asterisk-users] Streaming MoH on 1.4

Olivier wrote: > Hi, > > A somehow related question, is broadcasting streaming music as music > on hold, submitted to any licencing fee ? > > Regards > > > ___ > -- Bandwidth and Col

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