RE: [Asterisk-Users] Revieving some fax problems

I have run into a similar situation. One of our older faxes at the office seems to not work with spandsp module. The newer faxes work just fine. When I watch the logs, there appears to be communication from * requesting the fax to "slow down". When the fax machine does not respond, * seems to s

RE: [Asterisk-Users] is a dual 1.5Ghz server better than a single 3Ghz for a 100 Iax users asterisk server

One key that I have found is the more RAM the better. I am not discounting the CPU by any means and with the number of registrations you are talking about, I have not set up a system for that many concurrent users. I do have a 2x1.266 PIII w/ 2 Gigs of RAM that handles 75-85 concurrent SIP (GSM)

[Asterisk-Users] How to get names into the *411 directory

Here's how to get names in the directory: Even though there is some information here: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Directory http://www.voip-info.org/tiki-index.php?page=Asterisk+config+voicemail.conf It's not obvious how to make certain that there are names availabl

[Asterisk-Users] is a dual 1.5Ghz server better than a single 3Ghz for a 100 Iax users asterisk server

Hi everyone, I'm looking to buy a server that could handle 100 IAX users (g711)-(about 300 registrations) simultaneously. No zap channels. My budget is 1000$ us, Is a fast (3ghz) single server more reliable than a double cpu (like 1ghz) ? Will asterisk take full profit of two cpus? Isn't better

Re: [Asterisk-Users] Problems using SIPURA and MFC/R2

This could be a RTP sizing problem. try witth RTP 20ms in your sipuras. Regards. 2005/9/29, Flávio Eduardo de Andrade <[EMAIL PROTECTED]>: > > > > We are using MFC/R2 driver successfully in at least three places in Brazil. > I have problem with an Asterisk integrated with MFC/R2 with a Siemens Hi

[Asterisk-Users] Asterisk and RTP streams

Guys, I've been poking around trying to find a good answer for this via voip-info, google, etc... Haven't found anything that helps, so maybe you mates could.   A lot of my customers are using Linksys UAs (router/ATA PAP2) and some using Sipura SPA-2002s. Every once in a while, the customer

RE: [Asterisk-Users] WRT54GP2 SIP server on LAN port

I'm quite sorry mate, I didn't realize how you meant to connect it. I just recently tried it because I decided to connect my router back into the local network to find the system so that I could just have my * system take care of all calls, etc... I encountered the same problem. If you found any w

Re: [Asterisk-Users] CDR and TDS

Hi! use 0.62.x instead 0.63. http://lists.digium.com/pipermail/asterisk-dev/2005-April/011002.html Andy Kuo on 10/1/2005 07:01 wrote: > Hi, > > I'm trying to install FreeTDS. I followed the instructions on > http://www.voip-info.org/tiki-index.php?page=FreeTDS >

Re: [Asterisk-Users] Music on hold not initiating RTP stream?

On Fri, Sep 30, 2005 at 05:05:41PM -0700, Ray Van Dolson wrote: > I'll have to do some tcpdumps later to see what SIP messages are sent now when > I hit hold and what happens to the rtp streams. Did some briefly. No RTP stream initiates from the Asterisk server. When I hit hold on the phone, no

Re: [Asterisk-Users] OOH323C

Kanishka Somaratne wrote: hi has any one used OOH323C i tried this it is installed but do not know how to configure has any one used this, what is the best h323 addon to use with asterisk OOH323 has no jitterbuffer and does not work with cisco gw (incoming calls with g729). OH323 (latest ve

Re: [Asterisk-Users] Re: How can i call to a cellphone here in Mexico?

Hi all!: I finally have found the solution to the problem, it was on zapata.conf file, where i put txgain=25. Looks like 25 was too high, causing the cellphone company to redirect my calls. I've found that my limit is 11, if i choose a higher value for txgain, all the outgoing calls to cellular ar

Re: [Asterisk-Users] H323 and Asterisk

ooh323c installeed but do not know how to configure :( maybe googling or reading README can help woomera let me know if there's any one who has tried this. i've been testing this. It can do only alaw/ulaw and this is unusable for me. It works, but i've got some segfaults (using gnugk 2.2.

Re: [Asterisk-Users] Music on hold not initiating RTP stream?

On Fri, Sep 30, 2005 at 06:47:51PM -0500, Kevin P. Fleming wrote: > >The ATA's are Sipura SPA-2002's and I have MOH Server set to 899 on each. > > Take that out, you don't need it. > > None of this is needed; Asterisk will stream MOH to ATA 2 all by itself, > just by the fact that ATA 1 put ATA

Re: [Asterisk-Users] transfering calls, no ringing sent to caller

I'm still trying to get this to work. I tried downgrading my Cisco phones to an earlier SIP image from cisco, thinking that might be the problem. Not it. Currently running SIP 7.5. I noticed when I use the transfer feature, a ZOMBIE appears on the channel, and the caller I'm transfering an exten

Re: [Asterisk-Users] Music on hold not initiating RTP stream?

Ray Van Dolson wrote: The ATA's are Sipura SPA-2002's and I have MOH Server set to 899 on each. Take that out, you don't need it. However, with a call in progress, if I hit hold or flash on SIP ATA 1, it behaves correctly, but no music on hold is heard on SIP ATA 2. I can see in my Asterisk

[Asterisk-Users] Music on hold not initiating RTP stream?

I've been having problems getting MusicOnHold to work, so I've dumbed down my setup to as simple of a setup as I can. Asterisk 1.0.9. SIP ATA's (Sipura SPA-2002's) <---> <---> Both ATA's have public IP's. No NAT'ing going on here. Reinvites are allowed so the media stream bypases Asterisk

[Asterisk-Users] CDR and TDS

Hi,   I'm trying to install FreeTDS.  I followed the instructions on http://www.voip-info.org/tiki-index.php?page=FreeTDS , but still can't get it to work. I serched around trying to find instructions on it, and it seems the same info (even wording) appear on all sites I found.   I downloaded fre

Re: [Asterisk-Users] cisco phones problems

On 14:51, Fri 30 Sep 05, Edwin Lam wrote: > after much struggles. i've found out that if i ping the phone unit > from another computer constantly (couple pings every 5-10 sec) > the phone will operate fine. once i stopped the pings, the UNREACHABLE > message started to pop up and the drop calls pro

Re: [Asterisk-Users] Fritz, mISDN, Help

Derek Conniffe wrote: Hi again Jörg, The install_misdn Makefile doesn't seem to like my SMP "yes" in my kernel .config (it does a grep on the .config file, finds the line, and tells me) - I'm going to try the chan_misdn driver anyway and the server is an old HP netserver e800 which is a dual

Re: [Asterisk-Users] cisco phones problems

after much struggles. i've found out that if i ping the phone unit from another computer constantly (couple pings every 5-10 sec) the phone will operate fine. once i stopped the pings, the UNREACHABLE message started to pop up and the drop calls problems starts. seems like it's the firmware issue.

[Asterisk-Users] Fax with asterisk

Hi I did install-pdf to receive fax and configured AMP (Extension of fax machine for receiving faxes = system) to send it as an email attachment. However, when i try to send the fax, sending fax machine gets the * digital receptionist and doesn't accept the fax. it keeps prompting with options jus

Re: [Asterisk-Users] quick question on ztdummy

On 22:40, Fri 30 Sep 05, Tzafrir Cohen wrote: > On Fri, Sep 30, 2005 at 08:35:19PM +0200, Michiel van Baak wrote: > > On 13:28, Fri 30 Sep 05, Jerry Geis wrote: > > > asterisk is running fine. I removed my 2 port TDM02B and now I must > > > compile ztdummy. > > > I edited the makefile changed the

Re: [Asterisk-Users] SPA-841 "Decode Latency"?

> Does anyone have any familiarity with "decode latency," specifically > with Sipura devices? Why would it take 200+ms to decode a 20ms RTP > packet? G.711u has existed for over 30 years, how hard could it be? Although I have never seem the decode latency to go above 30 ms on a LAN, it does go up

Re: [Asterisk-Users] Revieving some fax problems

Try Hylafax, with external fax/modem, it works 99.999% It you try to route it via Asterisk (with NVFaxDetect) your success will be about 95% -- #Joseph On Fri, 2005-09-30 at 15:57 -0400, Alexandre Leclerc wrote: > Hi, > > We are recieving some faxes, but I would say that about 50% of them do >

[Asterisk-Users] SPA-841 "Decode Latency"?

We're investigating audio quality issues in our system; maybe someone can help. We're using Asterisk as a basic PBX, with a single PRI on one side and SIP phones on the other: Sipura SPA-841's. We're experiencing several audio effects which seem to commonly correspond to network failures (packet l

Re: [Asterisk-Users] 911 Q

On Fri, 2005-09-30 at 09:16 -0700, Ray Van Dolson wrote: > On Fri, Sep 30, 2005 at 10:20:12AM -0400, Joel Newkirk wrote: > > How can we achieve this, short of 'reciting' the unit number aloud at > > the beginning of the placed call? > > Hmm, could you just put the full address (including unit no.)

Re: [Asterisk-Users] quick question on ztdummy

On Fri, Sep 30, 2005 at 01:49:19PM -0500, Jerry Geis wrote: > On 13:28, Fri 30 Sep 05, Jerry Geis wrote: > >/ asterisk is running fine. I removed my 2 port TDM02B and now I must > />/ compile ztdummy. > />/ I edited the makefile changed the # ztdummy to just ztdummy (removed > #). />/ recompiled.

[Asterisk-Users] Revieving some fax problems

Hi, We are recieving some faxes, but I would say that about 50% of them do not work. We don't know why... is it something with the faxes speed, volume, etc? Should we use a real fax machine? Using a TDM13B with a rxgain of about 5.0... Thank you for any help. -- Alexandre Leclerc _

Re: [Asterisk-Users] quick question on ztdummy

On Fri, Sep 30, 2005 at 08:35:19PM +0200, Michiel van Baak wrote: > On 13:28, Fri 30 Sep 05, Jerry Geis wrote: > > asterisk is running fine. I removed my 2 port TDM02B and now I must > > compile ztdummy. > > I edited the makefile changed the # ztdummy to just ztdummy (removed #). > > recompiled.

[Asterisk-Users] Polycom IP301 Hangs on boot.

My Polycom IP301 hangs on "Processing Cfg..." Here is the boot log: 0930155446|so |4|00|-- Initial log entry -- 0930155446|so |4|00|+++ Note that bootrom log times are in GMT +++ 0930155446|wdog |4|00|Initial log entry 0930155446|cfg |4|00|Initial log entry 0930155446

Re: [Asterisk-Users] No ringback tone generated by Asterisk with OH323connections

No, I wasn't. I can't believe I made that stupid mistake. It started working after I added the call to Answer(). Thanks for your help. On Friday 30 September 2005 11:53, Brian C. Fertig wrote: > are you giving answer()? > > ..o---o.. > Brian

[Asterisk-Users] Linksys register hangs Asterisk!

Hey, I'w got a problem (bug maybe?). I have recently got my Asterisk to work perfect and I'm not trying to setup some dial routes and get the system working as I wan't it to. Yesterday I was installing Festival and also did a "aptitude upgrade" on my Debian Unstable installation. After that the

[Asterisk-Users] quick question on ztdummy

On 13:28, Fri 30 Sep 05, Jerry Geis wrote: / asterisk is running fine. I removed my 2 port TDM02B and now I must />/ compile ztdummy. />/ I edited the makefile changed the # ztdummy to just ztdummy (removed #). />/ recompiled. />/ did modprobe ztdummy and everything worked />/ />/ However

Re: [Asterisk-Users] Meet me conferencing without blind transfers ([EMAIL PROTECTED])

Mojo with Horan & Company, LLC wrote: rather than using the blind and attended transfer functions built into the phones, try asterisk's features.conf -- we use ** for attended and ## for blind. (always use blind for meetme conferences.) Thanks! Will try that! Jen _

[Asterisk-Users] voip alarm circuit

Hi, I've been trying to get my ADT alarm panel ( Lynx24R ) to work with asterisk and sipura SPA 2000. So far the alarm company has been recieveing lot of undefined events. Someone suggested I should change communication format to 4x2 instead of default Ademco CID. Unfortantely ADT is reluctant

Re: [Asterisk-Users] Maximum number of Digium Trunk Cards

Or buy a T1/E1 card. - Original Message - From: <[EMAIL PROTECTED]> To: Sent: Friday, September 30, 2005 9:41 AM Subject: [Asterisk-Users] Maximum number of Digium Trunk Cards > I've read several places that say you cannot have more than two 4 port Digium T1/E1 cards, as it would overl

Re: [Asterisk-Users] Any way to not overwrite sound files on compile?

On 9/29/05, Matt <[EMAIL PROTECTED]> wrote: > Every time I recompile Asterisk (or upgrade to a new CVS-HEAD, > whatever) asterisk overwrites custom files I have made. Granted, > these files are named the same as the asterisk default files > (vm-login.gsm, etc) because we had a person here record t

Re: [Asterisk-Users] Canada VOIP provider quality

When you say "sufficient capacity" what, exactly, do you mean? We monitor our B channel utilisation and add PRIs whenever we see peak usage above 90% (note -- peak, not average). We aim for less than 0.1% blocking factor and have not yet come close. Pretty cool. What tool do you use to mon

Re: [Asterisk-Users] CRITICAL PROBLEM

Actually, I ran into this problem as well. If progressinband=never (and that setting didn't really work until recently in chan_sip), then you won't get any media back from the FEMA number as they send their attendant message through early media with a 183/w SDP. So, asterisk gets the 183 and the

Re: [Asterisk-Users] Why does the s extension not work inmy extensions.conf file

But I thought s was start and so should not need to do this? Angus - Original Message - From: "Matt Riddell" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, September 30, 2005 1:55 PM Subject: Re: [Asterisk-Users] Why does the s extensi

Re: [Asterisk-Users] quick question on ztdummy

On 13:28, Fri 30 Sep 05, Jerry Geis wrote: > asterisk is running fine. I removed my 2 port TDM02B and now I must > compile ztdummy. > I edited the makefile changed the # ztdummy to just ztdummy (removed #). > recompiled. > did modprobe ztdummy and everything worked > > However, now when I re

[Asterisk-Users] quick question on ztdummy

asterisk is running fine. I removed my 2 port TDM02B and now I must compile ztdummy. I edited the makefile changed the # ztdummy to just ztdummy (removed #). recompiled. did modprobe ztdummy and everything worked However, now when I reboot it is not automatically loading ztdummy? Should it

[Asterisk-Users] Asterisk useable DID in Newmarket, Ontario

Does anyone know of a provider that a) allows/works using Asterisk b) provides local DIDs to the Newmarket/Aurora area? thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com h

[Asterisk-Users] Co-author of O'Reilly's Asterisk book presenting in Utah Valley

Anyone in the area (Utah Valley) is welcome to join us (UVLUG) at this free event. Besides being a great presentation, there will be plenty-o-swag (books etc..) If you need more info, this page has it: http://uvlug.org/modules/news/article.php?storyid=96 Hope to see you there! -- Gabriel Gunde

Re: [Asterisk-Users] Will a VIA Epia ME6000 with a 600MHz Eden fanless CPU be suffiecient for 8 extension system?

On Fri, Sep 30, 2005 at 01:32:07PM +0100, Angus Comber wrote: > Hello > > I am using a VIA Epia ME6000 with a 600MHz Eden Fanless CPU. Is this > likely to be enough power for a 8 extension system with 6 external pstn > lines? Probably, yes. > > How important is cpu? Is there some measure, e

[Asterisk-Users] SIP make outside call

Hi, I am can make local extension to and from SIP X-Lite softphone, but I can't dial out using X-Lite but local analog works just fine. Here are my conf files any idea? Thanks, David my sip.conf [general] bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=

[Asterisk-Users] C Manager Interface Client

List:   This is my first manager client that I've written so please bear with me:   I am trying to write a C manager interface client to interface with our CRM software. I am having an issue while reading the data from the manager interface.   I am writing this in C and I have the following cod

[Asterisk-Users] Maximum number of Digium Trunk Cards

I've read several places that say you cannot have more than two 4 port Digium T1/E1 cards, as it would overload the PCI bus. Is that true? If not, what is the maximum number of cards people are putting into boxes? If two cards is the limit, am I right in understanding that the preferred way t

[Asterisk-Users] Re: Correction: Asterisk sound files, audio bandwidth, and sound quality

In article <[EMAIL PROTECTED]>, <[EMAIL PROTECTED]> wrote: > > Now Digium hasn't made the standard prompts available in a format other > than gsm. I don't know why. I asked a couple of weeks ago, and Kevin said they no longer existed: http://lists.digium.com/pipermail/asterisk-users/2005-Sept

Re: [Asterisk-Users] CRITICAL PROBLEM

Tim McKee wrote: I'm running a large number (125) remote sip phones for FEMA in the Gulf area over satellite. I've run into a major problem and need some assistance. When dialing the FEMA voice response system, it appears that it never actually answers the phone. I never get audio when dialing

[Asterisk-Users] Re: No ringback tone generated by Asterisk with OH323 connections

In article <[EMAIL PROTECTED]>, Juan Jose Comellas <[EMAIL PROTECTED]> wrote: > I am using Asterisk (Debian unstable packages) with an OH323 connection to my > provider. Everything is working except for the generation of ringback tones > when I receive inbound calls from the PSTN. My provider tel

Re: [Asterisk-Users] CRITICAL PROBLEM

Tim McKee wrote: Is there any way to get SIP to pass audio prior to getting a call complete message? This is Asterisk CVS-HEAD 08-01-2004. There is another possibility... in your dialplan context that is handling the call from the SIP phone out to the PRI, issue an Answer() before placing t

Re: [Asterisk-Users] CRITICAL PROBLEM

Tim McKee wrote: Is there any way to get SIP to pass audio prior to getting a call complete message? This is Asterisk CVS-HEAD 08-01-2004. That's a pretty old version... But in any case, we are already working on the same issue (not with FEMA though) for another customer, we should have som

Re: [Asterisk-Users] Siemens TC35 GSM gateway

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Fri, 30 Sep 2005, Andrew Smith wrote: I have a TC35 and am keen to see if anyone has both voice and sms working from Asterisk through this device? Google tells me that a few people have theorised about it, I can't find anyone claiming to be doin

RE: [Asterisk-Users] OT: SIPSAK usage

Thanks for the reply. I am using the -H option to specify the IP address of the registrar, so no problem there. It would seem then that my port 5060 has to be explicitly set, which I *think* is under Advanced > Advanced Network > Network identity (port): - the default setting is blank. Would adding

Re: [Asterisk-Users] 911 Q

On Fri, Sep 30, 2005 at 10:20:12AM -0400, Joel Newkirk wrote: > How can we achieve this, short of 'reciting' the unit number aloud at > the beginning of the placed call? Hmm, could you just put the full address (including unit no.) in the E911 database for the corresponding numbers assigned? You

Re: [Asterisk-Users] zttest - 100% ?

Are you starting Asterisk with the -p option (high priority?) Also, do you get a different value if you run zttest this way: nice -n -20 zttest CarlosOn 9/30/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: Digium itself is saying their cards may work not properly with zttestresults below 99,98

[Asterisk-users]

I would integrated my Asterisk PBX with CRM software, and I tell you if you prefer Asterisk or [EMAIL PROTECTED] for programming to. Thanks ___ Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB http://mail.yahoo.it ___

Re: [Asterisk-Users] Any way to not overwrite sound files on compile?

Andrew Kohlsmith wrote: So wrap the install binary such that it checks for the existence first. Existence of what? The issue is that if we have a new version of a sound file, there's no way to know whether the one currently in place is 'original' or modified. Sounds tedious. Why not simpl

[Asterisk-Users] mISDN, HFC, W6692, one-way-voice problem

Hi All, I'm trying to use a HFC chip ISDN modem with mISDN and chan_misdn. The card is configured to NT, PmP mode. One Siemens ISDN phone connected to the modem. When I call the ISDN phone is called everything is just fine, but when I call from the ISDN phone I face some problems. - There is no

[Asterisk-Users] CRITICAL PROBLEM

I'm running a large number (125) remote sip phones for FEMA in the Gulf area over satellite. I've run into a major problem and need some assistance. When dialing the FEMA voice response system, it appears that it never actually answers the phone. I never get audio when dialing via SIP through a

Re: [Asterisk-Users] Correction: Asterisk sound files, audio bandwidth, and sound quality

Hi, Steve: Thanks for your comments! [EMAIL PROTECTED] wrote: > The recorded prompts supplied with Asterisk are encoded with the .gsm > codec. That makes them sound like audio sounds on your GSM cellphone. > Which is noticably worse than true PCM audio. > > Now in the telephone world "best q

Re: [Asterisk-Users] Re:Any way to not overwrite sound files on compile?

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Matt wrote: > What about during an upgrade though? From one version to another where > you actually want to replace asterisk? I mean it isn't just a few > files (I don't think?) that get copied when you do make install is it? > Aren't there a fair am

Re: [Asterisk-Users] Any way to not overwrite sound files on compile?

Matt wrote: A post-install would be great (or I myself can write a script)... it isn't that big of a deal.. I just wanted to see if I was over looking something. Tagging the sound directory for a version would also be good but if there is no way (and I do understand the reasoning) then I ca

Re: [Asterisk-Users] Correction: Asterisk sound files, audio bandwidth, and sound quality

On Fri, 30 Sep 2005, Stephen Bosch wrote: > How ironic that Allison (the woman who did the Digium prompts) is > Canadian... Heh. Well, I didn't notice a prompt where she said "aboot", eh? I'll take my foot out my mouth now... Steve ___ --Bandwidt

Re: [Asterisk-Users] zttest - 100% ?

Digium itself is saying their cards may work not properly with zttest results below 99,98 the card itself is working the way that we can call out and receive calls, but we encountered massive echo-problems - sometimes more, sometimes less even on lines within the same phone-provider and be sure

Re: [Asterisk-Users] Any way to not overwrite sound files on compile?

On Friday 30 September 2005 11:39, Kevin P. Fleming wrote: > That is correct. 'make install' installs the standard sound files along > with the binaries; if we did not do that, then when the code had been > changed to required new sound files they would not be present... So wrap the install binary

RE: [Asterisk-Users] OT: SIPSAK usage

snom phones by default do not accept SIP messages from other destinations that the registrar (in this case they send a error response) and they dont listen on port 5060 by default. Reason: SECURITY!!! If you want to lower your security, you can manually specify the SIP port to 5060 and manually di

[Asterisk-Users] It is possible to have 2 AVM Fritz! USB for multiple BRI access?

Hello Asterisk users, I would like to use 2 BRI lines on my * box but I haven't any PCI slot. Is it to possible to use 2 two AVM Fritz! USB. If not, what other solution can I use? Thanks Amaury ___ --Bandwidth and Colocation sponsored by Easynews.c

RE: [Asterisk-Users] G.729 patent in France

Steve Underwood [EMAIL PROTECTED] wrote: > A large percentage of the patents applicable to G.729 are held by France > Telecom. Now guess whether they bothered to get those patents in France. > British Telecom has a large number of patents in North America. It can't use its software-only patents in

[Asterisk-Users] Calls Dropping w/ Cisco 7960 Phones

Hello, I have scoured google for the last couple of days, implemented some changes but my issue is still occuring. My company uses a hardware Bridge System for conferencing. Typically, users will call in from cell phones but three always call from the VoIP system. Once or twice a day, one of the

Re: [Asterisk-Users] Any way to not overwrite sound files on compile?

A post-install would be great (or I myself can write a script)... it isn't that big of a deal.. I just wanted to see if I was over looking something. Tagging the sound directory for a version would also be good but if there is no way (and I do understand the reasoning) then I can just write a

Re: [Asterisk-Users] Re:Any way to not overwrite sound files on compile?

What about during an upgrade though? From one version to another where you actually want to replace asterisk? I mean it isn't just a few files (I don't think?) that get copied when you do make install is it? Aren't there a fair amount? On 9/30/05, Gustavo A. Gonzalez <[EMAIL PROTECTED]> wrote: >

Re: [Asterisk-Users] Any way to not overwrite sound files on compile?

Matt wrote: I end up with the version of Asterisk I wanted installed, my sound files get over written, and my config files stay in place =\ very odd and slightly frustraighting! That is correct. 'make install' installs the standard sound files along with the binaries; if we did not do t

[Asterisk-Users] Question about 3Com(r) 3101 Basic Phone

   Hi, i have one question, the 3Com® 3101 Basic Phone work with asterisk, if so i any a especial firmware o another thing. And wath other 3com ip phone product work with asterisk. I think is a good idea to create a list with the all voip device and the status with asterisk.    Thanks. ___

RE: [Asterisk-Users] Asterisk and telephone volume

Funny, I find it just fine, but I have had a few users complain about it. One lady said her phone wasn't working at all, so I checked it out, worked fine, and then she admitted to me that she was 80% deaf in her one ear, and on her old Vista 390 she had the volume cranked so it was ridiculously lou

Re: [Asterisk-Users] maximum retries exceeded on call

has somebody an advise. Do I need to provide more information? Regards Michael Michael Häberle wrote: Hi, I phone with phpagi and/or x-pro. Sometimes I get this warning in the asterisk-console: "maximum retries exceeded on call". I noticed when this message shows up, asterisk hangs up the call

[Asterisk-Users] strange wave like noise on sip handset

Hello On a Sipura SPA-841 handset (and also at other end) you hear a sea wave like sound - it gets louder then softer and continually repeats. I don't remember hearing this when using other handsets. But what is this effect? How can I reduce it? Angus ___

Re: [Asterisk-Users] G.729 patent in France

Amaury BOSSE a écrit : Hi all, I am building an Asterisk PBX with voicemail and music on hold functions. An ISDN BRI line will also be available and G.729 IP-phones will be used. Are there patents rights applicable to France? Which licence could I use and how many ones are required (o

Re: [Asterisk-Users] Canada VOIP provider quality

When you say "sufficient capacity" what, exactly, do you mean? We monitor our B channel utilisation and add PRIs whenever we see peak usage above 90% (note -- peak, not average). We aim for less than 0.1% blocking factor and have not yet come close. Pretty cool. What tool do you use to monit

[Asterisk-Users] Asterisk::AGI - What license ???

Hi, Asterisk::AGI is a fantastic piece of software. Unfortunately it comes with NO LICENSE WHATSOEVER. That's very annoying when you want to write GPL stuff that depends on it. I have tried mailing the author some time ago with no response. Does anybody know what the software license for thi

[Asterisk-Users] Asterisk and telephone volume

Hello I am using a Snom 190 and the quality seems OK. Trouble is the volume is quite low and full volume on the Snom is still too low. Is there something I can do on the asterisk to increase the volume? Angus ___ --Bandwidth and Colocation spo

RE: [Asterisk-Users] 911 Q

With hotel systems When some places a 911 call it is printed on the printer in the Front Desk, Hwen help arrives they usually go to the Frount Dsek anyway. I would set up a System() that would not only printout he romm number on the Front Desk Printer but also drop a call file in to trigger a cal

Re: [Asterisk-Users] No Incoming Calls on Asterisk

I use UNIVOICE provider, therefore you change sip.uni.it with your provider. View files ___ Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB http://mail.yahoo.it extensions.conf Description: 3949034846-extensions.con

RE: [Asterisk-Users] No ringback tone generated by Asterisk with OH323connections

are you giving answer()? ..o---o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan Jose Comellas Sent: Fri

Re: [Asterisk-Users] chan_zap.so ?

On Friday 30 September 2005 10:26, cyril SIMON wrote: > I've a little problem with my asterisk server. Yes, it is a little problem. > Today, I wanted to restart it and when I did it, my > asterisk server didn't want to start again. It's telling you the problem pretty damn clearly: > Sep 30 16:5

[Asterisk-Users] No Incoming Calls on Asterisk

Hi,   My VoIP service provider has provided me with a Sipura adapter and it works perefect. But I want to receive calls on my Asterisk server. I’ve tried everything but no success. I can dial successfully from Asterisk but it doesn’t receive calls. Dialing phone hears a busy tone and cell

Re: [Asterisk-Users] zttest - 100% ?

[EMAIL PROTECTED] wrote: just as an (bad) example: we are using an x206 and couldn't get the zttest above 99.975 equal what we were doing single irq, w/o acpi, w/o apic, different kernels, w/o hyperthreading, different slots, a.s.o. for an Digium wildcard TE110P so if

[Asterisk-Users] No ringback tone generated by Asterisk with OH323 connections

I am using Asterisk (Debian unstable packages) with an OH323 connection to my provider. Everything is working except for the generation of ringback tones when I receive inbound calls from the PSTN. My provider tells me that we're sending call progress indications and that because of this they're

[Asterisk-Users] chan_zap.so ?

Hi, I've a little problem with my asterisk server. I have managed an asterisk server for a few months one. Today, I wanted to restart it and when I did it, my asterisk server didn't want to start again. I looked at the various messages in /var/log and the same thing appears. " [chan_zap.so] =>

[Asterisk-Users] 911 Q

OK, got a question on 911. Looking into setting up a couple asterisk servers at a country club, with VOIP phones in each of 100 short-term residential rental units. Approx 100 extensions, approx 24 outside lines. Since everything is geographically at one location, reaching 911 correctly shouldn't

Re: [Asterisk-Users] Echo Cancellation not working in Zapata.conf

On Friday 30 September 2005 09:57, Alberto Risco wrote: > I have echocancel=yes in zapata.conf but when I do a zap show channel 1, > I notice echo cancellation is turned off. If the channel is not in use, echo cancellation will be off. Your show zap channel output shows it's on-hook, so the DSP a

RE: [Asterisk-Users] OT: SIPSAK usage

I'm using sipsak to send messages to Snoms in my subnet. At work, works fine: sipsak -M -O desktop -B "foo" -s sip:[EMAIL PROTECTED] -H 192.168.1.46 displays "foo" on the Snom display On my home LAN (AAH 1.5, Snom 190 3.60s, switched 100, no VLAN, no routing) the same command (modified for my LA

RE: [Asterisk-Users] Echo Cancellation not working in Zapata.conf

Nevermind.  It turns out that if you are not on an active call on the channel, the “zap show channel x” shows OFF by default.  After placing a call and checking the channel, it showed “Echo Cancellation: 128 taps, currently on” as it should.  So our setup is correct after all.     Albert

Re: Mathematicians wanted (was RE: [Asterisk-Users] Best echo canceller?)

Kris Boutilier wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andrew Kohlsmith Sent: Thursday, September 29, 2005 2:23 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Best echo canceller? On Thursday 29 September 2005 17:04, C

[Asterisk-Users] Echo Cancellation not working in Zapata.conf

I have echocancel=yes in zapata.conf but when I do a zap show channel 1, I notice echo cancellation is turned off.   I followed the article that talks about the order in which the statements need to be in zapata.conf to get echo canceling to work:     http://lists.digium.com/pipermail/

[Asterisk-Users] Not authenticate

Why Asterisk show this message? What I can do? Sep 30 15:45:18 NOTICE[3608]: chan_sip.c:9096 handle_response_invite: Failed to authenticate on INVITE to '"100" ;tag=as413bd6a8' -- SIP/sip.uni.it-df15 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Thanks

[Asterisk-Users] Not Authenticate

Why Asterisk show this message? What I can do? Sep 30 15:45:18 NOTICE[3608]: chan_sip.c:9096 handle_response_invite: Failed to authenticate on INVITE to '"100" ;tag=as413bd6a8' -- SIP/sip.uni.it-df15 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Thanks!!

Re: [Asterisk-Users] Best echo canceller?

On Friday 30 September 2005 08:57, Tzafrir Cohen wrote: > Is there a difference in cpu consumption? (which may translate to > latency if you have enough channels, I guess) No. it's just refactored and fixes a few variable inits and stuff IIRC. The patches on the bugtracker explained it quite we

[Asterisk-Users] TE410P not working

I'm trying to install a TE410P this is what happens with compiled zaptel 1.0.9, 1.2-beta and 1.0.9 from http://updates.xorcom.com/iso/ this is my zaptel.conf (checked with the provider the values): span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone=it defaultzone=it then I modprobe wc

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