Re: [asterisk-users] replace astdb with a cluster-capable sql database engine

To be perfectly honest, the REALTIME function is absolutely hideous when it comes to reading data from the RealTime database. What on earth the Asterisk developers were thinking when they replaced the perfectly usable RealTime (which sets a channel variable for each field in the database) with the

Re: [asterisk-users] Newbie Polycom: IP601 console with expansion module

Special dialplans for reception are entirely up to you. The only reason reception phones have different dialplans to normal extensions is that often people want the receptionist's phone to behave a little differently. The Polycom 601 (nor any of the other common IP phonse designed for receptionis

Re: [asterisk-users] queue log vs. cdr

Yes it is. The reason you get more entries in queue_log is that there are several queue_log events per call - most commonly you get an "ENTERQUEUE", "CONNECT" and "COMPLETECALLER/AGENT" for each call. Vieri wrote: > Hi, > > Surely, I must be overlooking something. If I run the > following SQL que

Re: [asterisk-users] voicemail and needed language to be selected

Not ideal if you've got people who speak multiple languages using the phone system. You may want to review http://www.voip-info.org/wiki/view/Asterisk+multi-language - I suspect this is going to do what you want it to. Alex Balashov wrote: Replace the vm-* recordings in /var/lib/asterisk/s

Re: [asterisk-users] Best alternative for getting prompts recorded.

Maybe the Alison voice is better, but I've found Cepstral to be a bit too mechanical. I'm using Ceptral Millie (since a UK accent is more acceptable in Australia than an American one) The best TTS engine I've /ever/ run in to was Nuance Realspeak (see http://www.nuance.com/realspeak/) though

Re: [asterisk-users] BLF and Snom phones

Bill Hackensack wrote: On Sat, Mar 22, 2008 at 7:17 AM, Philipp Kempgen <[EMAIL PROTECTED] > wrote: http://bugs.digium.com/view.php?id=5014 The response on that issue from Russell is the kind of response that really ticks me off. No, no, no, we don't

Re: [asterisk-users] How to capture destination number when receive call through ZAP

The only method I'm familiar with for an analogue line to signal which number was called is a very old service that loops the line first and then dials the number. The only way to capture this would be to handle the incoming line as a standard extension with a different context. I've only run

Re: [asterisk-users] How to capture destination number when receive call through ZAP

channel = Zap/1-1") in new stack It still does not give me the dialed number. Could you explain how to match it again the zap channel to extract the dialed number? Will I be able to get the dialed number if I am using a E1 line? Thanks, Mark On Mon, Mar 24, 2008 at 2:29 PM, Rob Hill

Re: [asterisk-users] How to capture destination number when receive call through ZAP

Distinctive ring is still not going to provide the line that was called in the ${EXTEN} variable, so you're still stuck with dialplan trickery to figure out which number was rung. Mojo with Horan & Company, LLC wrote: Distinctive Ringing might be available from your telecom provider. mark mo

Re: [asterisk-users] Menuselect?

Only if you're trying to compile Asterisk 1.2. Asterisk 1.4 also has the menuselect configuration, though for most applications you don't really need to fiddle with it. James M Kupernik wrote: There actually is no menuselect, its just a simple ./configure make make install in that order

Re: [asterisk-users] Asterisk not hanging up after voicemail

Most likely, you don't have any hangup detection available or configured. If these are analogue lines, you will almost certainly need to configure busy detection in order to figure out that the call has been terminated. Do some Googling for "asterisk busy detection" mark morreny wrote: Hi,

Re: [asterisk-users] SPA-962+ SPA-932- blf function

We have BLF buttons working fine on the SPA932 side-car. What does "show hints" tell you under Asterisk, and what syntax did you use when configuring the side-car buttons? John Meksavan wrote: Asterisk Users, I am running Asterisk 1.4.11 on Debian "Etch" system with the TDM03B wildcard.

Re: [asterisk-users] Newbie Polycom: DHCP/boot server supporting 2 models of phones

All Polycom phones use the same firmware and bootroms - one reason why the sip.ld is so damn large for them. Lee, John (Sydney) wrote: > I have a question about DHCP and boot server supporting more than 1 > model of Polycom phones. > > According to Polycom "standards", Polycom phone boots up to

Re: [asterisk-users] SPA-962+ SPA-932- blf function

Have you set a call limit for each SIP peer? This is now required as of version 1.4. It took me a while to figure out all the issues when migrating to 1.4. John Meksavan wrote: Thanks for you guys help. The status LED on the sidecar takes an awfully look time to change from GREEN to RED a

Re: [asterisk-users] interrupting MOH

You may be able to achieve the desired result using queues rather than Dial statements. Overkill perhaps, but it's the only way I can think to implement it at the moment. John Millican wrote: Tilghman Lesher wrote: On Tuesday 01 April 2008 05:14:25 Pete Kay wrote: I am hoping som

Re: [asterisk-users] Advice on best operator phone (with attendant console)

For a receptionist, you generally want to go with a quality phone since they're going to be the heaviest user of the phone system in the building. (Inbound/outbound call agents may take/make more calls, but their requirements are far more simple than the complex call juggling a receptionist ca

Re: [asterisk-users] Advice on best operator phone (with attendant console)

I'd find that very strange considering that the 57i itself has facility for at least 20 BLF buttons and /each/ attendant console has facility for another 60! Matt Watson wrote: We are using 57i + 560M combination as well... though we are not using the 57i ct... but the idea of giving them a

Re: [asterisk-users] Paging for analoge devices

Google is your friend. I discovered very quickly what they were talking about by googling. bilal ghayyad wrote: Dear Steve & Doug; Sorry I did not understand any thing from your reply. --- Bogen Rulez On 4/5/08, bilal ghayyad <[EMAIL PROTECTED]> wrote: Hi; Anyone

Re: [asterisk-users] Newbie Polycom: Where isSoundPointIPWelcome.wav used?

That would at least be long enough to cover the entire boot process. ;) Lee, John (Sydney) wrote: It's played at the completion of the boot process. It's always been very quiet on the models I've worked with. Thanks Erik. I can probably replace it with my beloved Mozart Symphony no 40 :

Re: [asterisk-users] Advice on best operator phone (with attendant console)

wrote: Guys thanks a lot. I should be going with a Polycom 650 for all such jobs. If grandstream receives such bad reviews- how are they selling anything? Phones hanging or voice cut-outs are simply unacceptable!! On Sun, 2008-04-06 at 14:12 +1000, Rob Hillis wrote: I'd find that ve

Re: [asterisk-users] Message waiting indication(MWI) for voicemail - to H323 endpoints

Alex Balashov wrote: Anisha Kumar wrote: Please provide the required Setup / comfiguration details or redirect to appropriate to resource. You are decreasing your chances of getting a favourable response with an imperative tone like that, which also suggests that you are categorica

Re: [asterisk-users] X-Lite and Presence?

Configure the extension as a softphone using the format @. Works fine for me - and works even better for agents! Simon wrote: > Hi There, > > We have some users using x-lite as their SIP phone... but im wondering > how to get the "Calls & Contacts" to show as being available (Or if it > can be

Re: [asterisk-users] X-Lite and Presence?

X-Lite. Of course, Asterisk will need a hint configured for that extension as well... Simon wrote: Thanks for the reply.. Sorry for the lame question.. Do i do that in X-Lite or Asterisk? On Wed, Apr 16, 2008 at 2:07 PM, Rob Hillis <[EMAIL PROTECTED]> wrote: Configure the extensio

Re: [asterisk-users] X-Lite and Presence?

them when someone closes their app. But not free/busy type changes.. Any idea why here? Simon On Wed, Apr 16, 2008 at 3:21 PM, Rob Hillis <[EMAIL PROTECTED]> wrote: X-Lite. Of course, Asterisk will need a hint configured for that extension as well... Simon wrote: Thanks for

Re: [asterisk-users] Parsing incoming extension till first @

Using _. is going to result in warnings. A much better practice is to use _X. Ali Jawad wrote: > Thx again patrick it worked, I used > > [google-in] > exten => _.,1,Set(dst=${CUT(EXTEN,@,1)}) > exten => _.,1,Dial(SIP/[EMAIL PROTECTED]) > > while it should have been > > [google-in] > exten => _.,

Re: [asterisk-users] how to copy a variable without interpretation of the content

Try TEST="${X-CALLID}"; and see how you go. Eric Dantie wrote: Sorry, bad expressed, what I want to know is how can I do this in AEL: I've already got a variable X-CALLID with the content ctprueba-123456789.12 How can I copy the content X-CALLID to the new variable TEST? something like TE

Re: [asterisk-users] Quality problems with ISDN PRI

Every CPU core shows up as a separate CPU under Linux. For those that have hyperthreaded processors, a single core processor will show up as two processors - assuming you have hyperthreading enabled. linuxian iandsd wrote: "top" says asterisk 1.2.25 is using multiple cores: Cpu0 :

Re: [asterisk-users] Quality problems with ISDN PRI

To the best of my knowledge, multi-core processors are not hyperthreaded - certainly my Core 2 Quad processor isn't. I would expect a Core 2 Duo to be the same. Steve Totaro wrote: On Thu, Apr 24, 2008 at 11:18 AM, Rob Hillis <[EMAIL PROTECTED]> wrote: Every CPU core sh

Re: [asterisk-users] Quality problems with ISDN PRI

Steve Totaro wrote: On Fri, Apr 25, 2008 at 11:10 AM, Doug Lytle <[EMAIL PROTECTED]> wrote: Steve Totaro wrote: > That is interesting. I have an intel C2D and I can only see two > procs, not four, is that normal? Are you sure what you are saying is > I believe Intel removed HyperThread

Re: [asterisk-users] Upgrading to 1.4

As is just about always the case, posting twice to the list within three hours is not only unlikely to get a faster response, I would in fact imagine it would /reduce/ your chances of getting a response at all. lotusscript wrote: A good while back when installing 1.2 there were major issues wi

Re: [asterisk-users] Quality problems with ISDN PRI

No, a dual core processor has two cores. :) My Quad core shows four processors. Steve Totaro wrote: On Sat, Apr 26, 2008 at 2:37 PM, Benny Amorsen <[EMAIL PROTECTED]> wrote: "Steve Totaro" <[EMAIL PROTECTED]> writes: > My dual proc, dual core AMD boxen show as four procs. I guess the A

Re: [asterisk-users] Newbie Queue: greetings when first joiningqueue

Lee, John (Sydney) wrote: Check the number of calls waiting in the queue, then play the message if more than 0 example code (written in the TBird IDE) Exten => 100,1,Answer() Exten => 100,n,Set(NumWaiting=${QUEUE_WAITING_COUNT(${QUEUENAME})}) Exten => 100,n,GotoIf($[${NumWaiting} =

Re: [asterisk-users] Hyperthreading and multicore

Matt Florell wrote: > Also, I have heard HT processors explained this way, on an HT > processor it's like running 2 virtual processors at 70% of the specs > of the processor with HT turned off. It's not really like that in all > situations, but overall it has held pretty much true for me in most >

Re: [asterisk-users] Digium Card: Power Connector, from SATA to NORMAL

Steve Totaro wrote: Man that is an ugly hack, but I guess it may be required in some rare situation, I guess if the power supply of the server is already nearing it's rating. I would still go with a sata to molex connector and even a Y molex splitter if at all possible. Actually I can see e

Re: [asterisk-users] Asending or Round robin with trunks sip

SIP channels can't be grouped. What you need when you're dialling is the following if you want to use all three SIP channels:- Dial(SIP/troncal-1/${number}) Dial(SIP/troncal-2/${number}) Dial(SIP/troncal-3/${number}) That way if the first Dial fails, it will try the second one and so on. Walte

Re: [asterisk-users] Digium Card: Power Connector, from SATA to NORMAL

Steve Totaro wrote: >> Actually I can see exactly where this may be useful. I can think of at >> least one customer off the top of my head who has insisted on having a >> TDM2400 installed in a Dell server that we know can't provide enough power >> to the card where it's being used as a collect

Re: [asterisk-users] UK BT ISDN30e PRI Problem

Probably for the best - you'd look mighty silly otherwise. (spoken by someone who's done his own share of jumping around, yellng "YES!") Steve Totaro wrote: > I never kick myself on issues like this. I enjoy the challenge and > the eventual success by jumping around and yelling "YES, YES, YES!

Re: [asterisk-users] Zap Channels Collide (Incoming & Outgoing)

There are krone blocks designed for CAT5, and I've seen them in use as well. However, there's no way I'd be using them for today's networks. /Especially/ having seen one of these krone blocks used to double-punch two network ports together. Bill Andersen wrote: > Oh, yes. I saw an entire Ca

Re: [asterisk-users] BLF Compatible Phones

SPA942s do not currently support BLF keys. The four "lit" buttons are line keys only with the current firmware, although our Linksys rep has assured us that it's a feature to be supported soon. John Signorello wrote: > We use the linksys 942's and they work flawlessly and are easy to setup > >

Re: [asterisk-users] Digium Card: Power Connector, from SATA to NORMAL

At this stage, I've only seen one machine that didn't come with the old style power connectors. SATA power connectors may be a standard, but they haven't (yet?) supplanted the older power connectors. In fact, most power supplies I've bought recently have had more molex style connectors than SA

Re: [asterisk-users] T38 fax solution with Asterisk possible?

mark morreny wrote: > Hi, > > I am looking for a very low cost way of receiving and sending T38 fax > reliably. Is there any possible solution using Asterisk as the PSTN > SIP gateay and Digium E1/T1 card? Is there other open source package > that can help to accomplish this purpose? Asterisk

Re: [asterisk-users] T38 fax solution with Asterisk possible?

Some. Apparently not complete - and not capable of acting as a gateway. Olivier wrote: I thought that 1.6 carried along T.38 origination-termination capabilities. Is it true ? 2008/5/22 Rob Hillis <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>>: mark morreny wro

Re: [asterisk-users] H.323 video support

Remind me to pick on your poor Spanish next time I see you for a mid-morning meal. :) Steve Totaro wrote: > On Fri, May 23, 2008 at 4:05 AM, Diego Moreno <[EMAIL PROTECTED]> wrote: > > When and where is the 1.6 brunch? ;-) > ___ -- Bandwidth a

Re: [asterisk-users] Dear asterisk-users@lists.digium.com May 80% 0FF

Steve Totaro wrote: Darn, it was 87% off just yesterday! But with all that wonderful value-adding spam, it's worth paying more for isn't it? (then again, I guess that very much depends on /what/ you're paying for!) ___ -- Bandwidth and Colocatio

Re: [asterisk-users] Asterisk 1.6 vs 1.4?

Brent Davidson wrote: > We're currently using Asterisk 1.4.19, Zaptel 1.4.10, > Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones. Why on earth are you running two layers of echo cancellation - hardware and software? To be honest, I think this is asking for trouble - I've seen two occasion

Re: [asterisk-users] Asterisk 1.6 vs 1.4?

Tzafrir Cohen wrote: > On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote: > >> Why on earth are you running two layers of echo cancellation - hardware >> and software? To be honest, I think this is asking for trouble - I've >> seen two occasions where h

Re: [asterisk-users] init.d script no longer uses safe_asterisk

I believe Ubuntu is in the process of migrating from sysvinit to Upstart. Upstart is supposed to be capable of monitoring services to ensure they don't fail, so I suspect this is likely to be the reason behind the safe_asterisk script not being used. Paul Belanger wrote: > I noticed safe_aste

Re: [asterisk-users] Asterisk 1.6 vs 1.4?

Tzafrir Cohen wrote: > On Thu, Jun 05, 2008 at 03:40:14AM +1000, Rob Hillis wrote: > >>> If you use a hardware EC (or technically: a span-specific echo >>> cancellation method) the generic Zaptel echo canceller (software-based, >>> OSLEC in this case) will

Re: [asterisk-users] Bad ringback tone on zap channel

In my experience, the ringback you get over a zap channel (be it analogue or digital) is generated by the remote end, /not/ Zaptel. The ringback you get over a SIP or IAX2 channel is often generated by either Asterisk or the SIP/IAX2 device you're calling from. James Lamanna wrote: > Hi, > I'v

Re: [asterisk-users] PoE budget

Chris Bagnall wrote: >> have used many fsm7326p to power 24 phones or 726tp to power 12 >> phones and they work great >> > > On the Linksys side, we have a load of SRW-224P switches out in the wild > powering 24 Snom 370s (around 7W each) off each switch. > > > Likewise, we sell these thi

Re: [asterisk-users] PoE budget

that holds true for /any/ switched network. Jerry Jones wrote: > On Jun 7, 2008, at 9:51 AM, Rob Hillis wrote: > > >>> On the Linksys side, we have a load of SRW-224P switches out in >>> the wild powering 24 Snom 370s (around 7W each) off each switch. >>> >

Re: [asterisk-users] Bad ringback tone on zap channel

u have a faulty PSTN line. James Lamanna wrote: > Hmm ok. > This was a call from a SIP phone registered with Asterisk outbound on > a Zap trunk. > So would Asterisk or the phone be generating the ringback tone in that case? > > It also happens very intermittently (maybe 1 in 10 ca

Re: [asterisk-users] SIP call, updated with CID as it becomes available

Steve Totaro wrote: > If you ever have problems with a call dropping after 30 seconds, > Answer() is usually the cause. > > Answer is the /cause/? Or do you mean it's the solution? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.c

Re: [asterisk-users] Idiot's Question

core show function SPRINTF does work on my 1.4.20 system. Eric "ManxPower" Wieling wrote: > Oddly "core show function SPRINTF" works on my 1.6. SPRINTF function > does not seem to be in 1.2 and I don't have any 1.4 systems. > > Venefax wrote: > >> Believe it or not, I cannot find online a sin

Re: [asterisk-users] need ata suggestion

IMO, yes - sort of. :) Since Linksys bought Sipura, you're probably looking at the Linksys PAP2 - the functional equivalent of the Sipura SPA-2000. They look different (better if you ask me - the LEDs are far better placed and more useful than they were on the Sipura units) but are pretty mu

Re: [asterisk-users] GXW 4108 asterisk configuration

Doug wrote: > There is a bug in these units that won't let > you put punctuation in the extension name. A Grandstream product with a bug... what an unusual concept. Seriously, with all the grief I've had with GXP-2000s, BT-200s and GXV-3000s, I wouldn't touch Grandstream gear with a barge po

Re: [asterisk-users] need ata suggestion

ond port needs to act like a seperate > line tied to the same DID in a hunt group. > > Eric > > On Tue, Jun 17, 2008 at 3:52 AM, Rob Hillis <[EMAIL PROTECTED] > <mailto:[EMAIL PROTECTED]>> wrote: > > IMO, yes - sort of. :) Since Linksys bought Sipura, yo

Re: [asterisk-users] GXW 4108 asterisk configuration

se a closed TCP session. I opened > a ticket with them and after a week their answer is . "use udp". > > Rob Hillis wrote: > >> Doug wrote: >> >>> There is a bug in these units that won't let >>> you put punctuation in the e

Re: [asterisk-users] Recommendations for Motel Instalation.

80 rooms? I guess you and I have slightly differing opinions as to what a "small" motel is. :) If you have 80 analogue channels, then you'd need 4 TDM2400P cards. Unless your server is powered by a small nuclear reactor, you'll be better off with either 3 E1 or 4 T1 channels banks or 2 32 po

Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

One of the biggest barriers to upgrading are the number of little "gotchas" in syntax changes that can make an upgrade from 1.2 to 1.4 quite painful. After the pain I went through upgrading to 1.4, I've always been recommending to people to think twice about upgrading if 1.2 does what they require

Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

You've hit the nail on the head with the crux of the pain I went through. Finding stuff that was broke that I didn't realise was broke until someone bothered to tell me about it. I'm sure everyone is familiar with just how often users report problems caused by themselves, but don't report stuff c

Re: [asterisk-users] AsteriskNOW release date???

Speaking of attacks that aren't fair. Trixbox != FreePBX. They're completely separate products. Tilghman Lesher wrote: >> FreePBX seems to be the most logical choice to me. >> > > Which is being leveraged to take away business to anyone who has not sworn > allegiance to Fonality. Sorry, co

Re: [asterisk-users] Asterisk 1.4 Fax

Unless your provider provides a T.38 gateway, fax over SIP is pretty much guaranteed to be unusable. Often you can get away with it over a LAN using G711a or G711u, but any of the lower bandwidth codecs /won't/ be able to properly handle fax calls. Whilst I haven't used it myself, I believe IAXmo

Re: [asterisk-users] Asterisk 1.4 Fax

Well that answers that question. I see that t38modem provides an H232 modem - is this unsuitable for HylaFAX's purpose? (ignoring the fact that it requires a kernel recompile on most newer distros.) Steve Underwood wrote: > Rob Hillis wrote: > >> Last time I heard IAXModem di

Re: [asterisk-users] Asterisk 1.4 Fax

Then I suggest you prepare yourself for a lot of pain. Fax over the 'net without T.38 is almost guaranteed to not work. Al lists wrote: > I'm not looking at T.38 , at this time its terminating a SIP trunk > with multiple DID's for fax. > I'm using this configuration with linksys PAP ATA and sati

Re: [asterisk-users] Asterisk 1.4 Fax

I'd say "consider yourself very lucky". I know I did some testing here some time ago with faxing over VoIP. * One extension to another over G711a with both extensions on the same LAN - worked 95% of the time * One extension on my Asterisk server to an Extension on a friend's A

Re: [asterisk-users] Asterisk 1.4 Fax

year. > > > > Jonn > > > > > > *From:* [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] *On Behalf Of *Rob Hillis > *Sent:* Tuesday, January 01, 2008 4:13 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Su

Re: [asterisk-users] Trixbox and mail2fax

Apparently not. I'm sure as heck not going to get involved in this argument again! :) Bill Hackensack wrote: > Do people even read the mail list anymore, or do they just land on > this planet, subscribe to the list, and ask the same questions that's > been asked over and over and over and over a

Re: [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk

The reason that IAX2 is considered good for NAT issues is that it uses only one port for both control messages and voice traffic as opposed to SIP that uses a predictable port for control messages and an unpredictable one for voice/video traffic. If both servers are behind NAT servers, you /will/

Re: [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk

it's router. > The first asterisk (client) wont need any port forwarding. > > Tim. > On 2 Jan 2008, at 10:18, Rob Hillis wrote: > > >> The reason that IAX2 is considered good for NAT issues is that it >> uses only one port for both control messages and voice tra

Re: [asterisk-users] iP0020 Phone busy signal all the time.

If you post console logs as suggested, someone might be able to offer further suggestions. William Herrera wrote: > Sip show peers will show the phone connected. > The phone display shows connected with IP, date and time and ... network > Icon not blinking. > If I access the phone through the web

Re: [asterisk-users] iP0020 Phone busy signal all the time.

sy signal at the phone end. > > > > WH > > > > > > *From:* [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] *On Behalf Of *Rob Hillis > *Sent:* Saturday, January 05, 2008 2:36 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:*

Re: [asterisk-users] Two Asterisk Boxes Playing Together

Google is your friend. http://www.google.com.au/search?hl=en&client=firefox-a&rls=org.mozilla%3Aen-US%3Aofficial&hs=jR6&q=asterisk+iax+two+servers&btnG=Search&meta= Shane D wrote: > Okay, here's the dal. > > Me and my friend both have asterisk boxes. I want to be able to type > extension 27 on my

Re: [asterisk-users] asterisk-1.2.26.tar.gz Thoughts?

I would suspect that your hardware is the cause of your problems. Running a production PBX system on a discarded desktop system is a /really/ bad idea. I would seriously consider an upgrade to your hardware. Ira wrote: > At 12:34 PM 1/18/2008, you wrote: > > >>> Although for some of us, or a

Re: [asterisk-users] asterisk-1.2.26.tar.gz Thoughts?

What you run it on is very much a function of how reliable you want the system to be. The better the hardware, the more reliable it will be. If you're running in a business environment, then I wouldn't recommend anything less than server grade - even if it's low end server grade. The company

Re: [asterisk-users] asterisk-1.2.26.tar.gz Thoughts?

er hardware. The real mistake is in putting the older hardware into full production. Tzafrir Cohen wrote: > On Sat, Jan 19, 2008 at 06:21:15PM +1100, Rob Hillis wrote: > >> I would suspect that your hardware is the cause of your problems. >> Running a production PBX system on

Re: [asterisk-users] asterisk-1.2.26.tar.gz Thoughts?

PC's age and when they age, things tend to go wrong, particularly when you upgrade software. Unusual crashes are usually the first sign that something is going wrong. To me, it sounds like you've put the money into many of the right areas - segregating your voice and data networks, going with dec

Re: [asterisk-users] asterisk-1.2.26.tar.gz Thoughts?

Not the first time I've seen something like this happen. If you read what I said, I wasn't saying that this /was/ what was happening with his hardware, merely that it's the first sign. Tzafrir Cohen wrote: > On Sun, Jan 20, 2008 at 06:33:31PM +1100, Rob Hillis wrote: >

Re: [asterisk-users] Qsig link

Pretty easy actually. - zaptel.conf span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 zapata.conf -- usecallerid=yes hidecallerid=no echocancel=yes echocancelwhenbridged=yes echotraining=800 relaxdtmf=yes rxgain=-3.0 t

Re: [asterisk-users] Your "favorite" Asterisk application.

Yes, but I already knew that. :) Paul Hales wrote: > I love writing dialplan, using vi. > > Does that make me weird? > > PaulH > > > On Wed, 2008-01-23 at 23:57 -0500, Ken D'Ambrosio wrote: > >> Hi, all. I've done some Asterisk recelling, but recently got roped into a >> Sr. SysAdmin position

Re: [asterisk-users] Your "favorite" Asterisk application.

Asterisk really comes into it's own with cute scripts that can do almost anything with ridiculous ease. One of the things I've done with a number of Asterisk machines is to put in a script that downloads the latest weather forecast and reads it back to you using a TTS engine. Ken D'Ambrosio wr

Re: [asterisk-users] Your "favorite" Asterisk application.

You know full well I'm not related to you - I just work with you. :) Paul Hales wrote: > With comments like that people are going to think that we aren't > related. > > PaulH > > > On Thu, 2008-01-24 at 16:46 +1100, Rob Hillis wrote: > >> Yes, but I

Re: [asterisk-users] autoprovision 200+ linksys phones setup

Hi Eric, You may want to contact me off-list - the company I work for offers a product which aims to be a zero configuration service for Asterisk. The Linksys 942 and 962 phones /are/ supported. Erick Perez wrote: > Hi there, > We have plans to install an office (not call center) with the follow

Re: [asterisk-users] Maybe a little OT---USB Handset

I doubt that chan_oss/chan_alsa directly support echo cancelling. However depending on exactly how you are using the inputs and outputs on a sound card, you could very well need echo cancellation. 90% of the time, echo is generated at the junction between a channel that separates received and tra

Re: [asterisk-users] Zaptel timer on Intel Dual Core servers

Likewise here. The company I work for sells duo core boxes (though mostly /with/ E1 cards) and we have no issue with timing. Chris Bagnall wrote: >> My question is if anyone else have seen this and if anyone has a >> possible solution? >> > > Nearly all of the boxes we've built over the las

Re: [asterisk-users] PRI with 20 channels

Leave your zaptel config as it is. In zapata.conf, configure your channel group as being channels 1-15,16-20. It does no harm at all to configure all the PRI channels in zaptel.conf. Since Asterisk refers to zapata.conf when it comes to actual utilisation of the channels, that's where you reduce

Re: [asterisk-users] voicemail to non-default context user does not work

According to voip-info, the syntax for the VoiceMail command is as follows... VoiceMail([/flags/]/[EMAIL PROTECTED]&[EMAIL PROTECTED]&boxnumber3]/) If you check the syntax for the VoiceMail command, it indicates that the mailbox parameter is /not/ optional, so I'm surprised this works at all. A

Re: [asterisk-users] Dialing SIP server user extension... Dial string issue...

Why are you specifying the password and server IP in the dial string when it's included in sip.conf? It's unnecessary. I believe that Dial(SIP/gs102/1234) will achieve what you want. ast guy wrote: > Hi, > > I'm trying to call a SIP server while providing the SIP server > username/password in d

Re: [asterisk-users] Dialing SIP server user extension... Dial string issue...

nf. I have all sip > buddies in Database. so will that work in this scenario ? > -ag > > On Feb 10, 2008 11:55 AM, Rob Hillis <[EMAIL PROTECTED] > <mailto:[EMAIL PROTECTED]>> wrote: > > Why are you specifying the password and server IP in the dial > string when i

[asterisk-users] Realtime SIP peers - reloading cached info

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi guys, I've been working on a little dialplan fragment for roaming extensions, however the customer wants us to set the MWI indicator for the roaming extension that has just logged in. We're using MySQL realtime, so I've figured out that RealTim

Re: [asterisk-users] Realtime SIP peers - reloading cached info

port and executes the asterisk commands. You execute asterisk command via agi not using system command > > -ag > > On Feb 11, 2008 11:24 AM, Rob Hillis <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: > > Hi guys, > > I've been working on a little dialp

Re: [asterisk-users] Grandstream GXP2000 Loses Connectivity

One thing to keep in mind is that the Grandstream's firmware is notoriously buggy and unreliable. I've got one GXP2000 here that is on the 1.1.5.15 firmware, and I wouldn't even consider upgrading other phones to them. Unfortunately, the quality of the Grandstream firmware is appalling and doesn'

Re: [asterisk-users] Realtime SIP peers - reloading cached info

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 If it is being removed in 1.6, I'm a little concerned since there's no mention of this when you show the application, nor on voip-info.org. What application/function is it being replaced by? Atis Lezdins wrote: | On 2/13/08, Rob Hill

Re: [asterisk-users] Realtime SIP peers - reloading cached info

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Atis Lezdins wrote: | By RealTimeUpdate do you mean func_realtime? It shouldn't care, as | cache is not implemented in realtime level, but higher (chan_sip). | | Are you sure you need "sip show XXX load". If you "sip prune" peer | data, it should be re

Re: [asterisk-users] Realtime SIP peers - reloading cached info

load" to re-cache the details. Works nicely, but it seems a bit ugly to me. Frankly, I'm surprised that RealTimeUpdate doesn't contain an option to flush and reload details, which would negate the need to employ other "hacks" to achieve this. Atis Lezdins wr

Re: [asterisk-users] Realtime SIP peers - reloading cached info

1.6 and schedule it for removal in 1.8. Half a development cycle isn't a very long time for a warning that a function will be removed. Atis Lezdins wrote: On 2/13/08, Rob Hillis <[EMAIL PROTECTED]> wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 If it is being removed

Re: [asterisk-users] Realtime SIP peers - reloading cached info

Johansson Olle E wrote: > So please rememner that there are a few independent regular Asterisk > developers out there that is not on the Digium payroll and still take > part in decisions about Asterisk. > Point taken. > Over a year is a long time for a warning like this, considering that

Re: [asterisk-users] Attendant phone

As far as I'm aware, only the Aastra 57i with three 560M modules would come close to your requirements. The 57i can display up to 5 extensions at one time with a further 15 being available by the use of multiple pages. The 560M modules can display up to 20 extensions at one time with three pag

Re: [asterisk-users] 57iCT BLF problem

I guess we ought to add "...beyond downgrading the firmware to 2.0.2" to that. :) Paul Hales wrote: > > We upgrade 2 of our Aastra 57iCT to the latest firmware (2.1.2.30) and > the BLF indicators no longer function. > > Has anyone had a similar issue? And a solution? > > PaulH > > > > __

Re: [asterisk-users] Digium stopped TDM400P production: alternatives??

The cards themselves are okay, but the extra level of configuration is a pain in the proverbial. Zaptel is already double-configured in both zaptel.conf and zapata.conf (that's not a complaint - I understand the reason for the separation) but the Sangoma cards require a /third/ level of configurat

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