Re: [Asterisk-Users] I love you!

1 solution 4 yout workstation: Linux Linux Linux Linux Linux Linux Linux Linux !!! :-) On Tue, 2004-05-11 at 05:53, John Fraizer wrote: > tmpm wrote: > > > Of course, and I suggest a firewall as well, but its NOT going to do > > anything for a purloined email some infected machine in Bumsq

[Asterisk-Users] * INSTRUCTIONS FOR MEMBERS OF THE COMMUNITY * Please read

Welcome to the Asterisk users community! Asterisk.org is a fast moving project. New code is added every day. Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Our community is also growing fas

[Asterisk-Users] AGI.pm wait_for_digit() not working for me!!!

Hello everybody!!!   I really need your help guys, I am using the AGI mode in meetme application, and  I want that AGI should wait for an input from the client/user i.e. a digit and then proceed, but I have used that AGI function agi->wait_for_digit(), but no usemy agi just passes, or

Re: [Asterisk-Users] I love you!

I LOVE YOU GUYS TOO!!! ASTERISK ROCKS!!! And as always the appropriate reponse seems to be wear a condom! :P bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

Re: [Asterisk-Users] Notice for Gentoo Users In Regard to mpg123

> I did see your post a couple days ago. Actually I didn't mention, but I was > using 0.59r but found it has problems dealing with variations between MP3 > files. What this really means is if you have a batch of MP3 files with a > mixture of 80K, 128K 256K etc then mpg123r will play the files

Re: [Asterisk-Users] Notice for Gentoo Users In Regard to mpg123

Thanks Brain, I did see your post a couple days ago. Actually I didn't mention, but I was using 0.59r but found it has problems dealing with variations between MP3 files. What this really means is if you have a batch of MP3 files with a mixture of 80K, 128K 256K etc then mpg123r will play th

Re: [Asterisk-Users] Notice for Gentoo Users In Regard to mpg123

> at that time, reverting to 0.59r (potential security issues) or > resampling all mp3 moh to the actual recommended 8khz seemed to be the > only workarounds. Heap-based buffer overflow in readstring of httpget.c http://www.mpg123.de/mpg123/mpg123-0.59r.tar.gz I doubt the file straight from the

Re: [Asterisk-Users] Notice for Gentoo Users In Regard to mpg123

Thanks for the effort but please read the list next time. (I'm the one that started the thread about mpg123 0.59r) bkw - Original Message - From: "David Liu" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, May 10, 2004 10:26 PM Subject: [Asterisk-Users] Notice for Gentoo User

Re: [Asterisk-Users] Notice for Gentoo Users In Regard to mpg123

I noticed this ~3 months ago ^_^ at that time, reverting to 0.59r (potential security issues) or resampling all mp3 moh to the actual recommended 8khz seemed to be the only workarounds. but now I am using 0.59s-r3 for a few weeks (masked as ~ in portage) and it's pretty much back to the previo

[Asterisk-Users] Notice for Gentoo Users In Regard to mpg123

Hi there, After scratching my head for days on why there is horrible white noise in musiconhold, I finally found out that the latest emerge version of mpg123: 0.59s-r2 is not working well with asterisk. However, if you get the latest source from mpg123.de mpg123-pre0.59s.tar.gz. Then it works wo

Re: [Asterisk-Users] I love you!

tmpm wrote: Of course, and I suggest a firewall as well, but its NOT going to do anything for a purloined email some infected machine in Bumsquatialand sending it as you. Theres only so much you can do. If you run your own mailserver, I suggest the following to keep your braindead users from b

Re: [Asterisk-Users] Problem with SMP?

Hi Brain, Yeah I know there is a Stable 1.0 version but unfortunately, Primus' SIP Proxy doesn't work with nicely with the SIP INVITE message sent out by the 0.9.0 or above versions...so I am stuck! David - Original Message - From: "brian k. west" <[EMAIL PROTECTED]> To: <[EMAIL PROTECT

RE: [Asterisk-Users] DNS load-balancing & SRV records

If only it were that easy. :-) In the situation you've described, you would successfully achieve load distribution but not necessarily load balancing and potentially confuse yourself to no end. This really is an issue of semantics in one sense, load balancing vs load distribution. If all serv

[Asterisk-Users] How do I catch someone pressing the * key?

I would like to be able to detect when someone dials *. What I'd like to be able to do is exten => *,1,Answer and catch it when the caller pressed the * key. Thanks! Paul Mahler [EMAIL PROTECTED] Signate, LLC PO Box 60430 Palo Alto,

RE: [Asterisk-Users] Callerid via PRI

I just created a patch to return the specific sequence number of calling number entries so you can ignore my plea for help. Thanks. -- Matthew Billings | Affordable WWW & Internet Solutions foreThought.net | for Small Business [EMAIL PROTECTED] | 910 16th Stre

[Asterisk-Users] +5 seconds delay when Switching to macros

I switched sections of my extensions.conf to macros and noticed a 5 second delay. has anybody experienced the same? how can I work around it? Regards, Uriel

[Asterisk-Users] Terrible TICKING sound

i'm getting a tick every second or so on all my calls. All channels are zap channels. Does anyone know how to fix this? Thanks! Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training

Re: [Asterisk-Users] Transfering with Grandstream Phones

I don't know about new firmware, I'm using 1.0.4.55, but the transfer works fine with gs BT-100 with cvs asterisk (Downloaded yesterday) Where can I get firmware 1.0.4.63? can somebody give me a link? and what improvements from firmware 1.0.4.55? Isianto > > Interesting! Because a few months ag

Re: [Asterisk-Users] ztcfg and Aastra 390 phones

This is the output: [EMAIL PROTECTED] root]# ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXO Kewlstart (Defaul

Re: [Asterisk-Users] ztcfg and Aastra 390 phones

run ztcfg -vvv bkw - Original Message - From: "Ronald R. McDaniel" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, May 10, 2004 7:37 PM Subject: [Asterisk-Users] ztcfg and Aastra 390 phones > > > I am wandering if anyone can explain this unusual scenario. I am running > Aster

[Asterisk-Users] ztcfg and Aastra 390 phones

I am wandering if anyone can explain this unusual scenario. I am running Asterisk on Fedora. After the box boots, I run insmod wcfxs at a command line. The phones keep the red message light on and are in a state like they are not recognized yet. If I then run the command ztcfg, I don't get an

[Asterisk-Users] CISCO 30 VIP and 12 SP+

I am trying to connect the above 2 models to Asterisk server. The firmware seems to be 2.02 and 2.04. The 30 VIP is connecting to the network with DHCP. If anyone is using these phones, can you please give me some help. Thanks, Naren _

Re: [Asterisk-Users] Low Bit Rate Codecs

Hi, DTMF will not pass reliably through *any* low bit rate codec, with the exception of some forms of ADPCM. GSM certainly does not work. When a VoIP system is configured properly it works just like a cellular telephone system, which suffers the same problem. It doesn't send the DTMF tones. It

RE: [Asterisk-Users] alternative FXO gateway to Mediatrix 1204?

Roughly $1000 - $1500 I believe (I can't get the exact number from this office). We got it from ABP Intl. (http://www.abptech.com/) who were very helpful. I put a review of our complete setup at http://www.voip-info.org/wiki-Asterisk+setup+success+5. --Ernest > -Original Message- > From:

Re: [Asterisk-Users] Virbiage FT201 IAX Hard Phone

We're waiting on the processor chip to be made for our first production run, there's currently no stock and they're in the process of making more. It's completely out of our hands and, trust me, I'm as frustrated as you guys are. As soon as our manufactures tell us the completion date, I'll pos

RE: [Asterisk-Users] alternative FXO gateway to Mediatrix 1204?

On Mon, 2004-05-10 at 12:37, Ernest W. Lessenger wrote: > We use an AudioCodes MP-108 and have been quite happy with it. NOTE: Make > sure you get the most recent software build, the one that came installed on > ours was REALLY old. Might if we ask roughly what you paid for it?

[Asterisk-Users] DNS load-balancing & SRV records

Let's say I have a third-party device acting as a sip<-->pstn gateway, a cluster of three asterisk servers, and a teensy bit of dns knowledge. Let's now say those asterisk servers are a1.company.com at 192.168.0.1, a2.company.com at 192.168.0.2, and a3.company.com at 192.168.0.3. 1. If I setup ro

[Asterisk-Users] polycom ip 500 registration problems

hello all, I'm having problems getting my polycom soundpoint ip 500 working, and was wondering if anyone would be willing to share their config files with me (the polycom configs). I have managed to get my boot server up and running, and the phone successfully updated its ROM, and downloaded t

Re: [Asterisk-Users] Asterisk on a dual processor machine

I had the exact same problem, updated kernel, problem fixed. - Original Message - From: "Scott Weis" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, May 10, 2004 4:24 PM Subject: Re: [Asterisk-Users] Asterisk on a dual processor machine > Try upgrading your kernel... This is

Re: [Asterisk-Users] Asterisk on a dual processor machine

Try upgrading your kernel... This is the original shipped kernel and is very buggy. I would suggest getting "yum" It can be retrieved from: http://ftp.freshrpms.net/pub/freshrpms/redhat/9/yum/yum-2.0.4-1.rh.fr.i386.rpm Once installed you can just type "yum update kernel" as root at a bash shell.

[Asterisk-Users] Asterisk on a dual processor machine

Hi there, i have a problem installing asterisk on a dual processor machine.I have Red Hat 9.0 with kernel-smp-2.4.20-6. I did the installation process with no problem, i used the asterisk stable version 1.0. The problem is that the machine has some troubles after Asterisk goes up, the CPU performa

Re: [Asterisk-Users] SIP in the UK

On Mon, May 10, 2004 at 08:58:23AM +0100, Gavin Hamill wrote: > http://www.voiptalk.org/ - this is the service-side of "TelAppliant", official > UK Digium resellers. > > I've written to VoIPTalk a couple of times and never got any response from > them, and their outbound calling rates aren't fa

Re: [Asterisk-Users] basic implimentation

Very good question. I suppose the only real *benefit* we can offer is more in-depth tech support as well as customized .conf files for a particular setup. Not everyone wants to learn * from soup to nuts, let alone the time investment, maybe they may just want an all in one, awesome phone system t

RE: [Asterisk-Users] I love you!

Of course, and I suggest a firewall as well, but its NOT going to do anything for a purloined email some infected machine in Bumsquatialand sending it as you. Theres only so much you can do. At 17:34 5/10/2004, you wrote: Hey All!! I know that viruses are inevitable, but I think that subscriber

Re: [Asterisk-Users] Question about Asterisk and its use

Perhaps point him to that beta2 users manual .pdf as well, its helped me greatly after some kind soul sent it to me when I was where he is now.. (I dont have the url handy, could someone cough it up please?) IMHO, its a very good place to get the overview, and then the wiki makes a lot more sens

Re: [Asterisk-Users] gui -apps

I'm working on a gtk dash board with similar features to the op_panel application. Ken Jones inter7.com On Monday 10 May 2004 2:52 pm, Christopher Wall wrote: > > Is anyone using Asterisk Flash Operator Panel? > > How well does it work? > > ___ > Aste

Re: [Asterisk-Users] I love you!

Getting them here as well, and from a lot of sources besides the * list. Seems theres a major outbreak of sendings of net-sky, bagel and a few un-identified ones. I dont know if they're related to the Sasser outbreak, (suspect in custody BTW) but they're allegedly targeting w2k systems especial

Re: [Asterisk-Users] Swap partition/file and Asterisk.

Hi! > Well the idea is to run it on a 2 * Opteron 242 with 4 GB of memory and > 4*73GB 15K-RPM disks RAID-1. However the systen should also be able to > accomodate up to 150 extentions with an approx. continous usage of 40% > (e.g. approx. 60 users simultanious using the phones), in such a setu

RE: [Asterisk-Users] basic implimentation

Do tell, what is the advantage of buying from you, rather than directly from Digium for the exact same price? It would appear that anyone with real interest in Asterisk would be choosing Digium as their hardware supplier, unless there was a *substantial* cost savings in buying elsewhere? -Ori

RE: [Asterisk-Users] Swap partition/file and Asterisk.

150? If you don't have to transcode you should get higher numbers.. if the media stream isn't going thru asterisk it bumps up WAY higher! bkw > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Jens P. Hansen > Sent: Monday, May 10, 2004

RE: [Asterisk-Users] I love you!

Hey All!! I know that viruses are inevitable, but I think that subscribers of this list do not deserve to be plagued with them. Please make sure that you are not sending viruses to other subscribers. You can do this via the following method. 1) run an online virus scanner to remove any viruses y

Re: [Asterisk-Users] Swap partition/file and Asterisk.

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 10 May 2004 04:54 pm, Jens P. Hansen wrote: > brian wrote: > >Why would it be? > > > >bkw > > Simply because of the theoretical latency which are implicit in > disk-swapping, when doing encoding/decoding within the Codecs. -e.g. as > VoIP are

RE: [Asterisk-Users] I love you!

I posted to the list instructions on how to remove a virus, but I don't See it. Hmmm. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Gallaway Sent: Monday, May 10, 2004 2:00 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] I love you! [EM

RE: [Asterisk-Users] Virbiage FT201 IAX Hard Phone

Ya ya last I looked all Uniden phones were h323 but this one is very new it seems! :P bkw > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Eric Wieling > Sent: Monday, May 10, 2004 4:18 PM > To: [EMAIL PROTECTED] > Subject: RE: [Aster

Re: [Asterisk-Users] Swap partition/file and Asterisk.

brian wrote: It hasn't ever been an issue for me.. but if you plan on running this on an under powered non-beefy box ie 1ghz 128 megs of ram. It might be wise not to setup swap. You can also run with -p bkw Well the idea is to run it on a 2 * Opteron 242 with 4 GB of memory and 4*73GB 15K-R

RE: [Asterisk-Users] Virbiage FT201 IAX Hard Phone

The UP200 is SIP. On Mon, 2004-05-10 at 14:17, brian wrote: > > The response from them less than 10 days ago was 'about 2 months'. Which > > is what they have been saying for the last *3* months. I'm giving up > > waiting. They seem to be vaporware. :( Does anyone have a supplier for > > the Unide

RE: [Asterisk-Users] SIP calls-per-second performance test tool

No h.323 is evil... PURE evil! :P bkw > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Philipp von Klitzing > Sent: Monday, May 10, 2004 4:09 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] SIP calls-per-second performance

Re: [Asterisk-Users] Re: cron job to reboot GS101

Brian McSpadden wrote: I have used the cron job before, and it worked fine, but didn't seem to be more than a hack to me. I found that if I turned off the "Subscribe for MWI" flag in the GS config page, and it stopped losing registration. Excuse me, but where do You find the "Subscribe for MWI

RE: [Asterisk-Users] Swap partition/file and Asterisk.

It hasn't ever been an issue for me.. but if you plan on running this on an under powered non-beefy box ie 1ghz 128 megs of ram. It might be wise not to setup swap. You can also run with -p bkw > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] O

Re: [Asterisk-Users] SIP calls-per-second performance test tool

> It seems that entire days pass by before I > hang up... very odd, and very counter-productive to getting good > Asterisk work done. Telephony is evil. P. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/ast

[Asterisk-Users] Callerid via PRI

When receiving multiple calling numbers via a PRI for a call setup, I cannot find the ability to select between either first or last. Is there a way to do this currently? If not, can anyone help me in getting this to work? Here is the dump of the PRI in intensive debugging. Please note that I wa

Re: [Asterisk-Users] Swap partition/file and Asterisk.

brian wrote: Why would it be? bkw Simply because of the theoretical latency which are implicit in disk-swapping, when doing encoding/decoding within the Codecs. -e.g. as VoIP are extreemly dependent on "realtime" convertions, it may be that a 2-300 ms. delay in disk-swaps may have a seriou

[Asterisk-Users] gui -apps

Is anyone using Asterisk Flash Operator Panel? How well does it work? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listi

[Asterisk-Users] wrong topic... sorry from now on gui apps

Sorry, I accidently replied to a post, rather than created a new one. Christopher Wall wrote: Is anyone using Asterisk Flash Operator Panel? How well does it work? brian wrote: Why would it be? bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]

Re: [Asterisk-Users] Swap partition/file and Asterisk.

Is anyone using Asterisk Flash Operator Panel? How well does it work? brian wrote: Why would it be? bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jens P. Hansen Sent: Monday, May 10, 2004 3:14 PM To: [EMAIL PROTECTED] Subject: [A

RE: [Asterisk-Users] SIP calls-per-second performance test tool

If you can only get it to compile! :P bkw > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of John Todd > Sent: Monday, May 10, 2004 3:16 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] SIP calls-per-second performance test tool >

RE: [Asterisk-Users] Swap partition/file and Asterisk.

Why would it be? bkw > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Jens P. Hansen > Sent: Monday, May 10, 2004 3:14 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Swap partition/file and Asterisk. > > I was wondering whethe

Re: [Asterisk-Users] Signalling C7 / SS7

That's the way to go: IMTs (or "SS7 trunks") for the bearer services and and SS7 for signaling. The cost benefits alone would make it a valuable project. The days of the huge end office switch are numbered. every building and the larger corporations will have its own softswitch. There's plenty of f

[Asterisk-Users] SIP calls-per-second performance test tool

http://sipp.sourceforge.net/ Anyone care to throw this at Asterisk to see what happens? I would, but I am having significant temporal shortfalls recently due to the apparent warping of the space/time continuum when I answer the phone with clients/associates. It seems that entire days pass by

Re: [Asterisk-Users] basic implimentation

yes, use the link posted below - Original Message - From: "Christopher Wall" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, May 10, 2004 11:51 AM Subject: Re: [Asterisk-Users] basic implimentation > when shopping for hardware, I have not found anything listed with the > verbag

[Asterisk-Users] Swap partition/file and Asterisk.

I was wondering whether swap-file/partitions is discouraged in Asterisk "production" environments ? K.Rgds Jens P. Hansen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update

RE: [Asterisk-Users] Virbiage FT201 IAX Hard Phone

Michael,   I’ve heard Todd’s name also, however a few people from the * IRC room gave him a call and were told that they would not be sold to.   I’ve had the same perception and experience with their staff. Extremely professional and go out of their way to make you happy.  A great compa

Re: [Asterisk-Users] Transfering with Grandstream Phones

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 10 May 2004 11:12 am, Jeremy Jones wrote: > The gs handytone 286 user manual at the gs website lists call > transfer/call forward as not yet implemented (the firmware version listed > on the manual is out of date, thoughg). I sent a query to

Re: [Asterisk-Users] basic implimentation

when shopping for hardware, I have not found anything listed with the verbage of bundle do you have a link to this? Steve Totaro wrote: If you just want two lines in and two analog phones then you can go with a Wildcard TDM400P (2 Port FXS + 2 Port FXO Bundle) You can buy it from a reliable

Re: [Asterisk-Users] Virbiage FT201 IAX Hard Phone

At 02:34 PM 5/10/2004 -0400, you wrote: The response from them less than 10 days ago was 'about 2 months'. Which is what they have been saying for the last *3* months. I'm giving up waiting. They seem to be vaporware. :( Does anyone have a supplier for the Uniden UIP200 phones? Tim -- This is

Re: [Asterisk-Users] German sound files available

ePyron Felix Deierlein wrote: But I am still not sure, where I sould place the german digits, letters and phonems. First I placed everything under sounds/de/.. but then digits did not work, then I linked it to /sounds/digits/de/ now I have german digits but saynumber is still english. I think you m

RE: [Asterisk-Users] Virbiage FT201 IAX Hard Phone

Thanks for the reply Tim. That's a real shame. The featureset of the soon-to-maybe-be Virbiage IAX hardphones seemed too good to be true. I guess I'll go ahead with a full SIP deployment, and hope IAX hardphones become a reality in the future. Brian D'Arcy -Original Message- From: [EM

Asunto: Re: Asunto: Re: [Asterisk-Users] Asterisk webmin

Ok .. tonight i test it and later i write with the results regards Ivan >-- Mensaje original -- >From: Bruce Ferrell <[EMAIL PROTECTED]> >To: [EMAIL PROTECTED] >Subject: Re: Asunto: Re: [Asterisk-Users] Asterisk webmin >Reply-To: [EMAIL PROTECTED] >Date: Sun, 09 May 2004 20:24:23 -0700 > > >Ju

RE: [Asterisk-Users] alternative FXO gateway to Mediatrix 1204?

> Can anyone recommend a FXO gateway product that does behave > in this "more > correct" manner? We use an AudioCodes MP-108 and have been quite happy with it. NOTE: Make sure you get the most recent software build, the one that came installed on ours was REALLY old. --Ernest > -Original M

[Asterisk-Users] Uniden UIP200 Review (Repost)

Hello Everyone, My company is about to deploy * as replacement for our existing commercial Altigen PBX. Meanwhile, I've been trying to find the best cost effective SIP VoIP phone which we can use in office for 20-30 employees, as well as a few remote staff. Normally I wouldn't post about a VoIP

RE: [Asterisk-Users] Virbiage FT201 IAX Hard Phone

Uniden does have a SIP phone. I posted a review of it on Friday of last week. There were some problems with the list last week, so look for a re-post of the review, as well as contact information for a distributor in a few moments. Brian D'Arcy -Original Message- From: [EMAIL PROTECTED]

Re: [Asterisk-Users] basic implimentation

If you just want two lines in and two analog phones then you can go with a Wildcard TDM400P (2 Port FXS + 2 Port FXO Bundle) You can buy it from a reliable merchant here: http://www.totarotechnologies.com/catalog/product_info.php/products_id/58?osCsid=f28a1614e1da27df9a5f1ad8bc832657 Remember FXS

Re: [Asterisk-Users] Question about Asterisk and its use

Perhaps consult an Asterisk consultant? I hear these guys are pretty good at helping and can point you in the right direction for a small fee ;-) Otherwise visit the wiki at www.voip-info.org and expect to spend a few weeks at a minimum sorting through it all like the rest of us did. - Origi

RE: [Asterisk-Users] app_sms - rocks!

Forgive my ignorance, but I believe this would be of benefit to many listening in. I'm taking it that SMS can be done with PRI, however I don't understand the specifics of it. I have a T100P card connected to a PRI - can I send/receive SMS message with it? I'm not having much luck, & I figure

RE: [Asterisk-Users] basic implimentation

Assuming that you have 1 <- analog lines <- 4 and that you want your phones to be 100% VoIP (i.e. no Analog handsets): You should just need the new Digium TDM04B bundle and the granstream phone(s). If you have 1 <- analog lines <= 2 and 1 <= analog phones <= 2 then you can use the TDM22B bundle.

RE: [Asterisk-Users] Virbiage FT201 IAX Hard Phone

> The response from them less than 10 days ago was 'about 2 months'. Which > is what they have been saying for the last *3* months. I'm giving up > waiting. They seem to be vaporware. :( Does anyone have a supplier for > the Uniden UIP200 phones? uniden's are all h323 last I checked. ( h.323 aka H

[Asterisk-Users] alternative FXO gateway to Mediatrix 1204?

I bought a couple of Mediatrix 1204's a few of months back. (Perceived advantages were relatively low overall cost and size per port, and it isn't nearly as vibration sensitive as a PC would be.) Rich Adamson's review from Feb 1 is comprehensive, and the only thing I'd like to add is this: One "

Re: [Asterisk-Users] Question about Asterisk and its use

Question number one is overwhelming... How do you want to use this product. That will dictate how you set it up. In any case, there is extensive documentation on this, please go there first and formulate more specific questions that we can help you with. Try here for starts: http://www.automated

Re: [Asterisk-Users] I love you!

[EMAIL PROTECTED] wrote: lovely, :-) Is it just me or where there allready 3 virus sent to this list today? Maybe time for denim to disallow attachments? :-) -- Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/l

[Asterisk-Users] basic implimentation

I have confused my self a little bit I want to have a couple of analog lines pointed at my asterisk box, and in turn piped out to a couple of phones. What cards do I need to get? Here is what I understand to be what I need in addition to my computer: Digium X100P + TDM400P Grandstream BT10

Re: [Asterisk-Users] Virbiage FT201 IAX Hard Phone

On Mon, May 10, 2004 at 10:53:14AM -0700, Brian D'Arcy wrote: > Does anyone have any recent news on the Virbiage FT201 IAX Hardphone? > I'd *really really* like to deploy these phones instead of SIP > hardphones, and I can't help but wonder if I'm going to shoot myself in > the foot (or another sen

Re: [Asterisk-Users] Signalling C7 / SS7

We are interested, too, but from a practical standpoint. We have IMT trunks from the LECs, and it would be great to be able to terminate them with Asterisk. That can't happen without SS7 support, so instead we can only terminate PRIs, which, of course, requires a switch that supports SS7. Bruce

[Asterisk-Users] Question about Asterisk and its use

Hello I am very new in this area, just start reading about Asterisk and VoIP 2 days ago. I am very interested in this product but not really getting good information on. Will appreciate of someone an answer these question in detail or direct me to right documents: 1. How to setup and use this Pr

Re: [Asterisk-Users] Voice Pulse and Incoming numbers problem

I just got an account with Voice Pulse and connected to them using IAX2. No problem at all with outgoing calls, however I can not receive any. After further investigation I discovered that the numbers they assinged to me were already in use!!! I am not getting much help from them, their suppo

[Asterisk-Users] Virbiage FT201 IAX Hard Phone

Does anyone have any recent news on the Virbiage FT201 IAX Hardphone? I'd *really really* like to deploy these phones instead of SIP hardphones, and I can't help but wonder if I'm going to shoot myself in the foot (or another sensitive area) by deploying a ton of SIP phones just to find the IAX Har

RE: [Asterisk-Users] SIP in the UK

I've had good experience with their support on the phone too. They know what they are talking about. P.s. Is this the correct way to reply to a thread? I read last week a complaint about how people reply for those with "threaded" viewers. -Original Message- From: [EMAIL PROTECTED] [

Re: [Asterisk-Users] SIP in the UK

At 08:58 AM 5/10/2004 +0100, Gavin Hamill wrote: http://www.voiptalk.org/ - this is the service-side of "TelAppliant", official UK Digium resellers. I've written to VoIPTalk a couple of times and never got any response from them, and their outbound calling rates aren't fantastic. I would be conc

Re: [Asterisk-Users] Signalling C7 / SS7

Storer, Darren wrote: Hi Roger, What hardware do you use to connect your asterisk box to a PSTN carrier via C7/SS7 (instead of ISDN PRI)? There is currently no native SS7 support within Asterisk. If your need is urgent you could source a protocol converter (PRI/C7) from a 3rd party like T

Re: [Asterisk-Users] Asterisk & Rhetorical Systems

So this integrates with asterisk? where do I find app_cepstral? Kyle - Original Message - From: "Ben Merrills" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, May 10, 2004 8:45 AM Subject: RE: [Asterisk-Users] Asterisk & Rhetorical Systems I took a look at their site and pla

RE: [Asterisk-Users] Asterisk & Rhetorical Systems

Yep and it worked OHHH so well too. :P bkw > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Andy Powell > Sent: Monday, May 10, 2004 10:33 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Asterisk & Rhetorical Systems >

RE: [Asterisk-Users] Low Bit Rate Codecs

> If you take into consideration the need to pass DTMF to/from various SIP > providers, how do they rank? > > I'd given up on g729 completely because none of the dtmfmode options > seemed > to work with g729 for several sip service providers. GSM seems to work > fine > in general. rfc2833 should

RE: [Asterisk-Users] Low Bit Rate Codecs

If you take into consideration the need to pass DTMF to/from various SIP providers, how do they rank? I'd given up on g729 completely because none of the dtmfmode options seemed to work with g729 for several sip service providers. GSM seems to work fine in general. Cheers, Rich > -Original

RE: [Asterisk-Users] Asterisk & Rhetorical Systems

I took a look at their site and played some of the demo's - can I have any comments from Asterisk users? How did they get on with it and what is the general opinion of the quality etc? This will be used for a major service line that reports faults and outages across a network. Kind Regards, Ben

RE: [Asterisk-Users] Re: X100P keeping PSTN line Offhook

I see that your line signalling is set to kewlstart... Are you sure that your telco provides this? Also, I found that I was having similar problems when there were other devices on the line (like fax machines). The problem usually occurred when someone tried to make an outgoing call on the same lin

RE: [Asterisk-Users] Signalling C7 / SS7

Hi Roger, > What hardware do you use to connect your asterisk > box to a PSTN carrier via C7/SS7 (instead of ISDN PRI)? There is currently no native SS7 support within Asterisk. If your need is urgent you could source a protocol converter (PRI/C7) from a 3rd party like Telesoft (Okeford 4000 etc.

RE: [Asterisk-Users] Transfering with Grandstream Phones

I'm talking about the ht, but I assume (big assumption, I know) that the sip stack is the same between the two. jeremy > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Dave Cotton > Sent: Monday, May 10, 2004 9:30 AM > To: [EMAIL PROTECTED] > Su

Re: [Asterisk-Users] Asterisk & Rhetorical Systems

hehehehhe Yes I know I use cepstral.. I wrote app_cepstral... (bkw messed with it too) Andy *** REPLY SEPARATOR *** On 10/05/2004 at 08:06 Eric Wieling wrote: >On Mon, 2004-05-10 at 05:37, Andy Powell wrote: >> I'd love to hear how you get on Ben, but I get the feeling that >R

[Asterisk-Users] Re: X100P keeping PSTN line Offhook

Tom, Rich and Atif, Regarding your responses, 1. I have previously tried the "callprogrees=no". Doesnt solve the problem. 2. If "busydetect=yes", calls to PSTN get droped in the middle of the conversations. 3. Havent looked into the MOH thingy. This feature has caused me other problems. Thinking of

RE: [Asterisk-Users] Transfering with Grandstream Phones

On Mon, 2004-05-10 at 09:12 -0600, Jeremy Jones wrote: > The gs handytone 286 user manual at the gs website lists call transfer/call > forward as not yet implemented (the firmware version listed on the manual is > out of date, thoughg). I sent a query to gs tech support a month or two ago > and re

[Asterisk-Users] Dropped calles (with mp3)

Hi I am still struggling with those dropped calls and am about to give up. I have created an MP3 of the exact scenario that I can reproduce when the call gets dropped. Maybe this might have an clue for somebody. http://atom.port11.net/media/disconnectasterisk.mp3 Also here is my zaptel.conf: [cha

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