[Asterisk-Users] Register replicaton and HA *

O yes :) So, for having a load balanced call administrarion. Using linux virtual server proyect I have run across a few design issues. The idea is easy. 3 servers. The Main server, to recive the calls, and the other 2 to originate the call. The load balancing of generating the call can be made

Re: [Asterisk-Users] Adit 600

I've just ordered an Adit 600 w/ 5xFXS cards and one CMG cards. As of my discussion with CarrierAccess, it seems to work great. I've also begun an configuration (mgcp.conf) until it arrives, and also there it seems to have great capabilities. There are alot of data sheets and information of all A

Re: [Asterisk-Users] IAX outgoing redundancy

On January 27, 2005 11:20 pm, Brian Dingman wrote: > To combat this problem you will want to change the following line to > actually do something: > exten => dial-NOANSWER,1,Hangup That's a *large* failure on LiveVoip's part, IMO. If I get a NOANSWER back I don't *want* to do anything -- there w

Re: [Asterisk-Users] I want to display my numbers for incoming calls when some one dials my number from any where

On Thu, 2005-01-27 at 23:15 -0800, Mazhar Hussain wrote: > Hi to all, > > I and using asterisk with following > > 1. TDM400p card with four FXS modules, > So there are four analog phone lines on four zap channels, > My setup is working fine. > And version is like such > Asterisk CVS-v1-0-11/27/0

[Asterisk-Users] I want to display my numbers for incoming calls when some one dials my number from any where

Hi to all, I and using asterisk with following 1. TDM400p card with four FXS modules, So there are four analog phone lines on four zap channels, My setup is working fine. And version is like such Asterisk CVS-v1-0-11/27/04-20:48:45 But when some dials form his number (suppose abc) to my number

Re: [Asterisk-Users] Bristuff ZapHFC and Loosing D-Channel

On Thu, 27 Jan 2005, Klaus-Peter Junghanns wrote: Am Donnerstag, den 27.01.2005, 16:01 +0100 schrieb Remco Barende: == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up that is the usual behaviour on a P2MP BRI line. When idle the telco will bring down layer 2 and layer 1. B

[Asterisk-Users] Caller ID in AU

Is anyone in AU successfully getting Caller ID from the analogue PSTN service? If so, what settings? -- Howard. LANNet Computing Associates; Your Linux people -- "When you just want a system that works, you choose Linux; when y

Re: [Asterisk-Users] Q: Can I over-ride the value of ${CALLERIDNAME} ?

On Fri, 2005-01-28 at 17:12, [EMAIL PROTECTED] wrote: > Folks, > > I'd like to change the value of ${CALLERIDNAME} for incoming PSTN > calls from certain numbers, but haven't found a way that works. The goal is > to provide more informative names on my phones' caller ID displays--e.g., I > w

[Asterisk-Users] Q: Can I over-ride the value of ${CALLERIDNAME} ?

Folks, I'd like to change the value of ${CALLERIDNAME} for incoming PSTN calls from certain numbers, but haven't found a way that works. The goal is to provide more informative names on my phones' caller ID displays--e.g., I would prefer to display "ROB CELL" instead of "CELLULAR CALL" whe

Re: [Asterisk-Users] POE and Snom220

Paul Hales wrote: We have a HP2626PWR powered switch we use for our VOIP switch. It will not power our new SNOM220's phones. It will power a lot of other phones we have, including a Snom200. http://www.snom.com/faq/FAQ-03-11-10-da.pdf says: snom220 fully supports 802.3af which is standardized Power

[Asterisk-Users] Dial and Macro Do not seem to be working

Hello, Here is the dial command: exten => 790,2,Dial(SIP/[EMAIL PROTECTED]|60|M(screen^${CALLERIDNUM})) Here is the macro [macro-screen] exten => s,1,Wait(0.2) exten => s,2,say number ${ARG1} exten => s,3,Read(ACCEPT|screen-accept|1) exten => s,4,GotoIf($[${ACCEPT} = 1 ] ?7:6) exten => s,5,SetVar(M

[Asterisk-Users] POE and Snom220

We have a HP2626PWR powered switch we use for our VOIP switch. It will not power our new SNOM220's phones. It will power a lot of other phones we have, including a Snom200. Any ideas? Or did Snom get something wrong? I would love an answer to this, as my boss has got a bit upset about the fact tha

Re: [Asterisk-Users] IAX outgoing redundancy

Just as an fyi.. one of the problems I am having with LiveVoip and my guess is that some of you are also is that the LiveVoip call starts making progress but for whatever reason it comes back and says nobody available. To combat this problem you will want to change the following line to actually d

RE: [Asterisk-Users] Trouble with Quicknet Linejack

My extensions.conf: [llamadaslocales] ignorepat => 9 exten => _915XX,1,Dial(Zap/1/${EXTEN:1},90,Tt) exten => _915XX,2,Congestion exten => _9XX,1,Dial(Zap/1/${EXTEN:1},90,Tt) exten => _9XX,2,Congestion [llamadas] exten => _3915XX,1,Dial(${FARMAR}/${EXTEN:[EMAIL PROTECTED]

[Asterisk-Users] SIP CANCEL problem

I am trying to configure Asterisk to receive an inbound call on a Zap channel T1 and Dial a SIP UA registered to Asterisk. SIP Debug and pcap output shows that asterisk is sending an INVITE, followed by an immediate SIP Cancel message. I hear one ring on the called party and then an immediate

Re: [Asterisk-Users] Stumped by BroadVoice SIP

> And need to have all of their four servers to listen to incoming > calls as ony one can send it in.. Why do you think that? The server you registered last with will send incoming calls. I've registered my several lines only at their LAX server for the last few months, and didn't miss a call. But

Re: [Asterisk-Users] Re: Polycom phones

Matt, you're lucky. They wasted about four months of my time before I finally gave up. John mattf wrote: Hello, When I talked with the VP of VOIP phone sales at Polycom about a year ago, he was offering a dedicated engineer for the Asterisk community that would work through issues like people ha

Re: [Asterisk-Users] OT: iax.cc/sixTel local DID question

Andrew Thompson wrote: > David Mallwitz wrote: > >>> Their isn't any indication of whether or not clicking the Add button >>> will immediately add a number to my account or take me to another screen >>> to pick a NXX. > > > >> The form lets you choose the NXX. > > > Actually, it didn't. > >

Re: [Asterisk-Users] OT: iax.cc/sixTel local DID question

David Mallwitz wrote: Their isn't any indication of whether or not clicking the Add button will immediately add a number to my account or take me to another screen to pick a NXX. The form lets you choose the NXX. Actually, it didn't. I asked in a ticket what happens and the response came back that

[Asterisk-Users] LiveVoip Expanded Codec Support & Feb Sale 1.2 Cents a Min USA & Canada

LiveVoip has expanded its Codec support to include IAX2  uLaw  G.723  G.729  GSM  iLBC   LiveVoip LLC Feb Calling Specials USA & Canada 1.2 CentsRates apply to purchases made after 8 p.m. EST Jan 27th.   Lower 48 USA 800 Rates Feb Special 1.2 cents a minute.Take Advantage of Volume Buying

Re: [Asterisk-Users] Sound quality tuning with VOIP/Grandstreams... echo, cut out, codecs, asterisk

What is VAD ? Voice Activity Detection -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users

Re: [Asterisk-Users] OT: iax.cc/sixTel local DID question

Andrew Thompson wrote: > When you choose to add an unlimited local DID to your account from their > control panel, do you get to pick the prefix/NXX, or just the area code? > Their isn't any indication of whether or not clicking the Add button > will immediately add a number to my account or take m

Re: [Asterisk-Users] Multi Asterisk Server Transfers

> Anyone know a good IAX phone (not softphone)? Only phones I know that support IAX can be found there : http://www.iaxtalk.com/index.php ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-us

Re: [Asterisk-Users] Sound quality tuning with VOIP/Grandstreams... echo, cut out, codecs, asterisk

PCMU is g711 ULAW and PCMA is G711 ALAW. ALAW is more common in Europe. Not sure about VAD. On Thu, 27 Jan 2005 18:44:09 -0700, Kim Lux <[EMAIL PROTECTED]> wrote: > > Thanks for the tips. > > The Grandstream doesn't have a G711 or uLaw option for codecs. It has > PCMU, PCMA and iLBC. Are any o

[Asterisk-Users] ChanIsAvail not working

I'm testing ChanIsAvail context and it is not working for me. exten => 55,1,ChanIsAvail(SIP/11&SIP/21) exten => 55,2,Cut(theChannel=AVAILCHAN,,1) exten => 55,3,Dial(${theChannel},r) exten => 55,4,Hangup exten => 55,102,Goto(s,4) It is not dialing SIP/21 when I'm talking on SIP/11, it execute Hang

Re: [Asterisk-Users] Sound quality tuning with VOIP/Grandstreams... echo, cut out, codecs, asterisk

Thanks for the tips. The Grandstream doesn't have a G711 or uLaw option for codecs. It has PCMU, PCMA and iLBC. Are any of these related to G711 ? Grandstreams have echo cancellation and it appears to be working after a few seconds of conversation. What is VAD ? On Thu, 2005-01-27 at 20:22

Re: [Asterisk-Users] Linux Bridge + QoS Shaper HOWTO available

Asterisk allows setting custom TOS setting, so perhaps you could pick a TOS precedence not normally used, such as CRITIC/ECP or Internetwork Control, in which case your TOS would be 0xB0 or 0xD0 respectively. On Fri, 2005-01-28 at 02:10 +0100, Jasper Spaans wrote: > That could be nice, however, t

Re: [Asterisk-Users] Sound quality tuning with VOIP/Grandstreams... echo, cut out, codecs, asterisk

Try using g711 (ulaw) and make sure to turn Silence Suppression OFF as asterisk needs the full audio stream for assembling the audio streams. Once you get the call quality good using g711 (ulaw), then you can play around with the other codecs (g729, etc). Also, try a 20ms frame size. Unfortunate

Re: [Asterisk-Users] Linux Bridge + QoS Shaper HOWTO available

That could be nice, however, there could be more traffic using tos 0x10. I wouldn't bet on using it in a config like this. On Thu, 27 Jan 2005 15:32:44 -0800, Mike Machado <[EMAIL PROTECTED]> wrote: > You could also change the tos setting in sip.conf and use a rule similar > to the one found in th

RE: [Asterisk-Users] Need some advises configuring asterisk to callover INTERNET

Hi, You might want to first read http://www.digium.com/handbook-draft.pdf which explains most of the basic stuff. Most of the questions you'll have will be answered on http://www.voip-info.org or by reading http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_ v1/docs-html/boo

Re: [Asterisk-Users] SIP + NAT = horrible mess

Comments below. On Fri, 2005-01-28 at 08:18 +0800, Leo Ann Boon wrote: > > Kim Lux wrote: > > >I was expecting to have to port forward too and yet our setup doesn't > >require it, not on the laptop nor on the wireless router. > > > >I think as long as the SIP clients open a port on the NATing

Re: [Asterisk-Users] SIP + NAT = horrible mess

Kim Lux wrote: I was expecting to have to port forward too and yet our setup doesn't require it, not on the laptop nor on the wireless router. I think as long as the SIP clients open a port on the NATing device and keep them open so the SIP provider can connect to it, all is well, even if STUN i

Re: [Asterisk-Users] Anyone having problems with LiveVoIP?

Glenn Powers wrote: I ordered an 800# from LiveVoIP two days ago. I can register with Asterisk just fine, but when I call my 800#, I get a fast busy. I emailed support a day and a half ago and have heard NOTHING from them. VoicePulse Connect and VoipJet both work great for me. Someone on -users

Re: [Asterisk-Users] Re: phone rings when I'm using it over VOIP - WHY?

[snip] > > My setup is really simple. > > I have Sipura-3000 connected to * with phone1 and another SIP phone2. > > Here is my context: > > exten => 1,1,Dial(${phone1},20,tr) > > exten => 1,102,Dial(${phone2},20,tr) > > > > I have setup two phones and have VOIP, when I make call over VOIP I > > th

[Asterisk-Users] OT: iax.cc/sixTel local DID question

When you choose to add an unlimited local DID to your account from their control panel, do you get to pick the prefix/NXX, or just the area code? Their isn't any indication of whether or not clicking the Add button will immediately add a number to my account or take me to another screen to pick

[Asterisk-Users] Problems making SIP URL outgoing dial

Hi, I'd like to call my friends through their SIP URLs. I've found two approaches for doing this in Asterisk: - one is to prepend some numbers at start and catch them - the rest of called string is used for SIP URL - another approach (that I like better) is to use catchall pattern at the end of c

[Asterisk-Users] Voicemail attachment not being emailed out

I am running [EMAIL PROTECTED] Voicemail works fine but does not email out the voicemail attachments. Any suggestion? --- Voicemail.conf [general] #include vm_general.inc #include vm_email.inc [default] 201 => {password},Jeff G Laptop,[EMAIL PROTECTED],,attach=ye

RE: [Asterisk-Users] Linux Bridge + QoS Shaper HOWTO available

You could also change the tos setting in sip.conf and use a rule similar to the one found in the LARTC. This is a bit easier, and perhaps cleaner than matching many ports. # tc filter add dev eth0 parent 1:0 prio 10 u32 \ match ip tos 0x10 0xff \ flowid 1:4 That would match TOS of 0x

RE: [Asterisk-Users] Tortoise CVS download for Asterisk Docs

- Install TortoiseCVS from the link provided - Inside the file explorer, right click where you want to check out the docs - click cvs checkout in the menu - in the CVSROOT field, enter :pserver:anonymous:@cvs.sourceforge.net:/cvsroot/asterisk (note the : before pserver AND after ano

Re: [Asterisk-Users] Trouble with Quicknet Linejack

Marcelo Echeverria wrote: I have a Quicknet Linejack in /dev/phone0. My phone.conf is: [interfaces] mode=dialtone format=slinear echocancel=medium context=mayores device => /dev/phone0 Only I can mark 7 digits, soon asterisk tries dial automatically. I cannot mark 8 or more digits. 6 or less di

[Asterisk-Users] res_python

Does anyone know how I can get res_python? The location http://vox.groovy.net/moin/PyAsterisk I got from the wiki seems to be down. Thanks, Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/as

[Asterisk-Users] Trouble with Quicknet Linejack

I have a Quicknet Linejack in /dev/phone0.   My phone.conf is:   [interfaces] mode=dialtone format=slinear echocancel=medium context=mayores device => /dev/phone0     Only I can mark  7 digits,  soon asterisk tries dial automatically. I cannot mark 8 or more digits.   6 or les

Re: [Asterisk-Users] SIP + NAT = horrible mess

I was expecting to have to port forward too and yet our setup doesn't require it, not on the laptop nor on the wireless router. I think as long as the SIP clients open a port on the NATing device and keep them open so the SIP provider can connect to it, all is well, even if STUN isn't used. I

Re: [Asterisk-Users] Re: Polycom Phones

- Original Message - Hmm. Your own web site has it priced between the 500 and 600. If the difference is good support versus zero support, wouldn't the $50 difference between the 500 and the 480i be saved in the first 20 minutes you spend fighting with a problem? Another factor is that one

Re: [Asterisk-Users] SIP + NAT = horrible mess

Jean-Michel Hiver wrote: Hi Guys, After days of fiddling, I can't really get my SIP device to work communicate with Asterisk behind NAT. Sometimes the STUN server is flaky, sometimes the device isn't reachable if the connection is dropped and then put back on, sometimes it registers OK, sometim

AW: [Asterisk-Users] Re: Making digital/data calls through asterisk

Hi Miguel! Im trying to use the method with the isdn-modem . If the TE410E arrives, I'm going to try it and put it in the wiki. I just gut zapras working at home, so that's working fine. Next step coming soon;-) Thanks, Andre > -Ursprüngliche Nachricht- > Von: [EMAIL PROTECTED] [mailto:

[Asterisk-Users] Asterisk CVS on FreeBSD-stable gmaking result

Hi everyone, I've just compiled Asterisk from CVS in my FreeBSD-stable machine. It went fine. I did had to customised the Makefile though: ASTLIBDIR=$(INSTALL_PREFIX)/usr/local/lib/asterisk ASTVARLIBDIR=$(INSTALL_PREFIX)/var/lib/asterisk ASTETCDIR=$(INSTALL_PREFIX)/usr/local/etc/asterisk ASTSPOO

RE: [Asterisk-Users] Linux Bridge + QoS Shaper HOWTO available

Does your script work for SIP too? Only for IAX?   Iâd like to put two of your bridges in my setup â one between the local LAN and the net, and the other at our branch office between the LAN and WAN there.   But I need to be able to allow for sip to be QOSed, as we currently have an ast

[Asterisk-Users] Asterisk auto-dial out deliver message

I create all my .call files first in a temp directory then run a PHP script from cron once a minute. The PHP script monitors the number of files in the outgoing directory and moves the .call files over from the temp directory a few at a time. This keeps the outgoing directory topped off with the

Re: [Asterisk-Users] SIP + NAT = horrible mess

Here is another way to approach the problem: if you are using SIP phones to connect to a SIP provider and are using * in between, try turning off * and setting the * computer up as a simple NATing server and make sure you don't have some sort of network issue that has nothing to do with *. Once

RE: [Asterisk-Users] iax.cc / sixtel are they legitimate?

I have one DID and make outgoing calls through Sixtel. Their support has been good with the minor issues I've seen. The're very * aware. I also use Voipjet for outbound and Clearpath for an 800 number. All provide IAX termination. All three companies have been great compared to my prior experienc

Re: [Asterisk-Users] SIP + NAT = horrible mess

On Thu, 2005-01-27 at 17:25, Kim Lux wrote: > On Thu, 2005-01-27 at 16:57 -0500, David Boyd wrote: > > Will you Please share your configuration, I was ready to give up, > > thinking no one had been successful. > > I am not using Asterisk, so I can only give you the Grandstream part of > things.

Re: [Asterisk-Users] Outgoing call on Zap channel always gives DIALSTATUS ANSWER even when BUSY?

Jeremy Lichfield wrote: Outgoing call on Zap channel always gives DIALSTATUS ANSWER even when BUSY? Has anyone run into this? If you are using ANALOG ports then this is correct. The telco does not signal the status of the call so Asterisk must assume it's always answered as soon as it's done di

RE: [Asterisk-Users] Linux Bridge + QoS Shaper HOWTO available

Does your script work for SIP too? Only for IAX?   I’d like to put two of your bridges in my setup – one between the local LAN and the net, and the other at our branch office between the LAN and WAN there.   But I need to be able to allow for sip to be QOSed, as we currently have an ast

Re: [Asterisk-Users] SIP + NAT = horrible mess

On Thu, 2005-01-27 at 16:57 -0500, David Boyd wrote: > Will you Please share your configuration, I was ready to give up, > thinking no one had been successful. I am not using Asterisk, so I can only give you the Grandstream part of things. Maybe some of the Grandstream parameters will twig an id

[Asterisk-Users] Outgoing call on Zap channel always gives DIALSTATUS ANSWER even when BUSY?

Outgoing call on Zap channel always gives DIALSTATUS ANSWER even when BUSY? Has anyone run into this? Here is my conf files: Zaptel: span=823,1,0,d4,ami e&m=1-24 loadzone = us defaultzone=us Zapata: usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewa

Re: [Asterisk-Users] Multi Asterisk Server Transfers

Philipp von Klitzing wrote: Instead I'd go for a co-located Asterisk that the remote SIP devices register with, and then link both * boxes (co-located and central office) using IAX2 with IAX native transfers enabled. Of course this means that the office * _only_ talks IAX and that all calls to t

RE: [Asterisk-Users] Stumped by BroadVoice SIP

I had a lot of problem with them to set up.. You need to register to sip.broadvoice.com And need to have all of their four servers to listen to incoming calls as ony one can send it in.. Just posted my config two days ago. http://lists.digium.com/pipermail/asterisk-users/2005-January/085736.htm

Re: [Asterisk-Users] Avoiding queue retries without hangs?

Bruno Hertz wrote: Just a point of order, there is no Asterisk 1.0.12. The latest is 1.0.5. Sure, sorry, it's actually 1.0, i.e. CVS-v1-0-12/18/04-22:40:47. Actually CVS does not have a "version". You are using the CVS 1.0 branch dated 12/18/04. You might consider upgrading to 1.0.5 release o

Re: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard ApplicationforAsterisk

Bob Goddard wrote: What's wrong with ODBC? Was this not discussed here or on the dev list? I have only used it once, but was not a complaint of it is that it is rather limited? I would always rather go native. Developers tend to ensure a native interface works first before any other. I'm loathe (

RE: [Asterisk-Users] iax.cc / sixtel are they legitimate?

Been using them successfully for a couple of months now. No minimum volume... And my volume is LOW. Besides, 800#s are free elsewhere, but these guys have better customer service than "elsewhere". > -Original Message- > From: Jon Gabrielson [mailto:[EMAIL PROTECTED] > Sent: Thursday, Ja

[Asterisk-Users] Channel Groups?

Is it possible to build grousp of channels? I have a series of extensions which are receiving incoming calls to various virtual organizations. Something like this... [foo-incoming] exten => 21, 1, Goto(ACorp|1000|1) exten => 212333, 1, Goto(BCorp|1000|1) . . . exten => _NXXNXX,

[Asterisk-Users] Stumped by BroadVoice SIP

Hello guys. I am a fairly new user to Asterisk, and I'm just having a tough time. My goal is to set up a VOIP PBX. I have signed up with a BroadVoice number, and I have three systems with SIP phones. The PBX and the SIP phones are all behind a Cisco PIX running NAT. I am using Asterisk CVS ve

Re: [Asterisk-Users] SIP + NAT = horrible mess

On Thu, 2005-01-27 at 16:06, Kim Lux wrote: > I've got Grandstreams (SIP devices) working behind double NATs, none the > less. > > I recommend turning STUN off and make sure that your SIP devices are > generating random port numbers. If they generate static port numbers, > you'll get port collis

Re: [Asterisk-Users] Moh in meetme doesn't work if I transfer tomeetme

Hi, - Original Message - From: <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, January 27, 2005 9:10 PM Subject: Re: [Asterisk-Users] Moh in meetme doesn't work if I transfer tomeetme > It's hard to tell without seeing your config files,

[Asterisk-Users] Re: Making digital/data calls through asterisk

There is the ITU V.150 that defines a framework for transmiting modem signals using VoIP, in a efficient way, but there is no implementation other that R&D in big companies. If you only want to forward an analog modem call over a PRI to the PSTN, you may need a channel bank connected to the TE4

RE: [Asterisk-Users] Multi Asterisk Server Transfers

Simple as that? Anyone know a good IAX phone (not softphone)? Thanks Mike >Then you need to use the same protocol to the provider. One office is >using SIP, the other is using IAX. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265

Re: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard ApplicationforAsterisk

On Thursday 27 January 2005 21:14, Brian Capouch wrote: > Bob Goddard wrote: > > No, everyone should go what ever they are more comfortable with. > > If the code does not have support for a certain database and a > > user would like it, then why not have it in? The database specific > > code should

[Asterisk-Users] Asterisk auto-dial out deliver message

It appears that my choices for auto-dial out are one-call-at-a-time or all-calls-at-once. Does anyone have a solution for calling 100 people, 10 at a time? thanks, glenn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digiu

RE: [Asterisk-Users] iax.cc / sixtel are they legitimate?

I have a couple of toll-free numbers from them. They are legit. I have had a few minor issues, but nothing I can't live with. --- Kelly D Griffin Network Engineer Tantella Wireless http://tantella.com 479.273.9992 Voice 479.464.8998 Fax -Original Message- From: [EMAIL PROTECTED] [mailt

Re: [Asterisk-Users] Avoiding queue retries without hangs?

> Just a point of order, there is no Asterisk 1.0.12. The latest is 1.0.5. Sure, sorry, it's actually 1.0, i.e. CVS-v1-0-12/18/04-22:40:47. Thanks, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailma

Re: [Asterisk-Users] Digium and Intel Chipset compatability

We're having problems with a HP DL360 G4. TE410p simply does not generate any interrupts. Digium tech support say that they will have some more information for me by the end of the week. Julian. Martijn van Oosterhout wrote: Hi, I'm going to be setting up some machines with 4 port E1 cards fro

RE: [Asterisk-Users] ASTCC Trunks

Thanks for this pointer, I'll try to do that. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe Sent: Thursday, January 27, 2005 5:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ASTCC Trunks Pres

RE: Re: [Asterisk-Users] Changing mailbox greeting {Scanned}

I over looked the "u" or "b" in front of the extension. Works great. Thanks, David On Thu, 2005-01-27 at 19:44 +0100, Michiel van Baak wrote: > > The unavailable msg is used when the voicemail app picks up after no phone > > on your extension picks up, ie you are not available to answer the ph

[Asterisk-Users] Making digital/data calls through asterisk

Hi! We planing to by some PRI/BRI equipment to replace our existing telephone system. So I am going to try this: ISDN Card Outgoing Digital Call / Capability: Unrestricted digital information -> octobri -> asterisk -> TE410E -> Internet Provider / Receiver for Capability Unrestricted digit

Re: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard ApplicationforAsterisk

Bob Goddard wrote: No, everyone should go what ever they are more comfortable with. If the code does not have support for a certain database and a user would like it, then why not have it in? The database specific code should be as abstract as possible to allow this, perhaps being written in its ow

Re: [Asterisk-Users] Multi Asterisk Server Transfers

Mike Sander wrote: I believe this is what I have, but it still insists on running the transfer from the head office. Example: Provider --- IAX --- Head Office Provider --- SIP --- Remote Office Provider --- PSTN (Provider is the same * server in all cases) Call comes from PSTN to Head office. Head

Re: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard ApplicationforAsterisk

Bob Goddard wrote: No, everyone should go what ever they are more comfortable with. If the code does not have support for a certain database and a user would like it, then why not have it in? The database specific code should be as abstract as possible to allow this, perhaps being written in its ow

Re: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard ApplicationforAsterisk

On Thursday 27 January 2005 20:08, Brian Capouch wrote: > Bob Goddard wrote: > >>ODBC is ultimately the way this stuff ought to all be coded. > > > > No it's not. > > Well that was sure a convincing argument :-) You started it! > The alternative seems to be nicely framed before us now in this ver

Re: [Asterisk-Users] SIP + NAT = horrible mess

I've got Grandstreams (SIP devices) working behind double NATs, none the less. I recommend turning STUN off and make sure that your SIP devices are generating random port numbers. If they generate static port numbers, you'll get port collisions. The other parameter to watch is the "keep alive"

RE: [Asterisk-Users] Multi Asterisk Server Transfers

I believe this is what I have, but it still insists on running the transfer from the head office. Example: Provider --- IAX --- Head Office Provider --- SIP --- Remote Office Provider --- PSTN (Provider is the same * server in all cases) Call comes from PSTN to Head office. Head office transfers

Re: [Asterisk-Users] Bristuff ZapHFC and Loosing D-Channel

Hi This is normal behaviour if you are running isdn in PTMP mode, at least in sweden. It is bothering with these messages, but I'we been running my test site with two PTMP lines for months and everything worked fine. It is said that some operators keep trying to turn off the D-channel in PTMP-mode

Re: [Asterisk-Users] Anyone having problems with LiveVoIP?

You must have us confused with another company as PayPal is not a form of payment that LiveVoip accepts. Brandon Patterson LiveVoip LLC > figured it wasn't worth PayPal'n them $200 for International. But then I saw > the support rep's response that it was user error and I wonder if that's is > t

[Asterisk-Users] Tortoise CVS download for Asterisk Docs

http://www.asteriskdocs.org/modules/tinycontent/index.php?id=4   Can I make a suggestion that some documentation is provided for the Tortoise CVS download of the asterisk docs. I’ve tried every combination and I cant get it to work.   I’m assuming it must work otherwise it wouldn’t hav

Re: [Asterisk-Users] Linux Bridge + QoS Shaper HOWTO available

I know that you can't really shape inbound traffic, but I think the killer shaper would start delaying TCP traffic as soon as total incoming bandwith utilization hits 75%. Unless somebody floods your link, it should work pretty effectively. Yeah that's what is going on in my scripts. I sacri

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 6, Issue 432

This is auto reply I am away from . ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-us

[Asterisk-Users] Digium and Intel Chipset compatability

Hi, I'm going to be setting up some machines with 4 port E1 cards from Digium and I'm being told that TE410 is incompatable with several Intel chipsets including the ones in a lot of Dell server systems. Is this true? I can't find any confirmed details on the mailing list about it. Also, the emai

RE: [Asterisk-Users] Re: Polycom and call waiting again..

True, this is only for the inbound side of the equation. I forgot about the outbound side. For that you can try the following example. [in-from-sip] exten => _X.,1,Cut(test=CHANNEL,/,2) exten => _X.,2,Cut(test=test,,1) exten => _X.,3,SetGroup(${test}) exten => _X.,4,Goto(default,${EXTEN},1) The

[Asterisk-Users] iax.cc / sixtel are they legitimate?

Does anyone have any experience with iax.cc/sixtel? Are they a legitimate company? From their website it looks like you can get a private incoming 800 number for 30 cents/month plus 2 cents/minute. Somehow that pricing seems a little cheap for a DID number. I assume there has to be some minim

[Asterisk-Users] SIP + NAT = horrible mess

Hi Guys, After days of fiddling, I can't really get my SIP device to work communicate with Asterisk behind NAT. Sometimes the STUN server is flaky, sometimes the device isn't reachable if the connection is dropped and then put back on, sometimes it registers OK, sometimes it doesn't, etc. I've

[Asterisk-Users] Bad ECHO problem after upgrade to HEAD version

I installed the HEAD version of Asterisk on January 24 and ever since then I have a really bad echo problem.   I have tried to go back to the stable version, but it won't install.   Anyone else having this trouble?  Suggestions?   Thanks, Brian   Here are what I believe to be the pertinent de

Re: [Asterisk-Users] Linux Bridge + QoS Shaper HOWTO available

[EMAIL PROTECTED] wrote: I've created a pretty complete HOWTO on creating a Linux Bridge (using Fedora) to shape LAN <--> WAN traffic. It includes installation instructions, a script to configure the bridge (which you install as a service), and 2 scripts to configure the network interfaces using

Re: [Asterisk-Users] Moh in meetme doesn't work if I transfer to meetme

It's hard to tell without seeing your config files, but ... your first trace show : -- Started music on hold, class 'default', on IAX2/[EMAIL PROTECTED]/1 while your second trace show : res_musiconhold.c:466 moh_alloc: No class: random So it looks like the context the tranfert is from as MOH cl

RE: [Asterisk-Users] Anyone having problems with LiveVoIP?

This thread sparked my interest in LiveVoip's International rates, but then this thread then scared me away from LiveVoIP's services. People mentioning the services not working properly, at one point it was blamed on a needed Asterisk patch and then it started working again w/o the patch, etc. so I

Re: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard ApplicationforAsterisk

Bob Goddard wrote: ODBC is ultimately the way this stuff ought to all be coded. No it's not. Well that was sure a convincing argument :-) The alternative seems to be nicely framed before us now in this very thread: everyone coding for hisher own favorite DB, flame wars about "you didn't pick th

Re: [Asterisk-Users] Avoiding queue retries without hangs?

Bruno Hertz wrote: Talking * 1.0.12 here. Just a point of order, there is no Asterisk 1.0.12. The latest is 1.0.5. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or u

RE: [Asterisk-Users] Anyone having problems with LiveVoIP?

I have a DID from Livevoip. It is a low usage DID. I have had no problem with support. I get a reply back within the hour on the couple time I had a problem. Jeff -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brian Dingman Sent: Thursday, January 27, 200

[Asterisk-Users] Festival Error

Just installed festival and I get this error in asterisk CLI   SIOD ERROR: wrong type of argument to car : wholeutt Jan 27 11:46:09 WARNING[25917]: app_festival.c:444 festival_exec: Festival returned ER     ___ Asterisk-Users mailing li

[Asterisk-Users] X100P/Zaptel on Gentoo Sparc64

Just wondering if anyone knows if a X100P/Clone would work on say a Sun Ultra 10 running Gentoo linux for sparc...? I've seen on voip-info.org that it won't run on solaris sparc, but nothing mentions gentoo(linux)/sparc...? ___ Asterisk-Users mailing lis

[Asterisk-Users] Linux Bridge + QoS Shaper HOWTO available

I've created a pretty complete HOWTO on creating a Linux Bridge (using Fedora) to shape LAN <--> WAN traffic. It includes installation instructions, a script to configure the bridge (which you install as a service), and 2 scripts to configure the network interfaces using traffic control. http://w

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