[Asterisk-Users] re:asterisk installation error

Tilghman, i am on Linux localhost.localdomain 2.4.18-3bigmem #1 SMP Thu Apr 18 07:17:10 EDT 2002 i686 unknown cm = Designs __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree

Re: [Asterisk-Users] Fax over SIP alaw/ulaw

At 21:20 16-11-2003 -0600, you wrote: You will need to check with Cisco to see if the ATA188 has the same issues with faxing as the ATA186. http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_field_notice09186a0080094af7.shtml Be advised this item regards the ATA186-L series, and the

Re: [Asterisk-Users] iax configuration

Hi, Thankx for the promt reply..was out of town,will get back to u with more. mukta From: Michael Van Donselaar <[EMAIL PROTECTED]>To: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] iax configurationDate: Fri, 14 Nov 2003 10:06:02 -0600Reply-To: [EMAIL PROTECTED]On Fri, 14 Nov 2003 05:36:25 +

RE: [Asterisk-Users] Call transfer

WARNING[1242952640]: File app_dial.c, Line 318 (wait_for_answer): Unable to forward voice This is what I get And a crash Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, 17 November 2003 5:14 PM To:

Re: [Asterisk-Users] Call transfer

Hi Mick, It's going to be hard for anybody here on the list to help you, unless you are more specific, ie, what you did exactly to get a crash, and console output (with verbose set) debugs, logs (under /var/log/asterisk) and some configuration files. We'll be in a better position to help you then

[Asterisk-Users] Updated Asterisk-NL

I have updated the voice prompts (mostly small fixes, but still work left to be done - however, it's usable now, I'd say) and the patch file (to current CVS) for Asterisk-in-Dutch. As far as I can tell, all of the grammar work is now in - if anyone has feedback to share, please do so before I

Re: [Asterisk-Users] ISDN debugging and SIP dial-in issue]

(I have some problems with my mailing-list alias, I hope this doesn't get sent twice) On Sat, Nov 15, 2003 at 04:35:20PM +0100, Philipp von Klitzing wrote: Thank you for your comments Philipp: - with a SIP phone configured as 192.168.1.190, and with its SIP server being

[Asterisk-Users] receive fax with isdn

I saw that it is possible to receive fax with asterisk. Does this only work with Zap or is it also possible to use ISDN (with isdn4linux) and how should one do this (cant find the RxFax app)? gr. Jordi ___ Asterisk-Users mailing list [EMAIL

[Asterisk-Users] mgcp audiocodes mp200

hi, i've successfully configured 2 audiocodes mp104 fxs and fxo configured using SIP with asterisk currently i want to add another 2 boxes of audiocodes mp200 to add up a total of 4 trunks how can i combine this mgcp to run together with asterisk SIP? Thanks in advanced Steven

[Asterisk-Users] asterisk iEnsemble H323 communication

Hi, Has any one tried out the Ensemble! VoIP Gatewaywith asterisk over H323? Any help would be appreciated. Regards,Shamly

RE: [Asterisk-Users] dtmfmode SIPDtmfMode

As far as I know I'm making sip calls: From the laptop I'm calling 999 with the following configuration sections: Part of extentions.conf: [sipCompany] exten = _XXX,1,SIPDtmfMode,inband exten = _999,2,VoicemailMain(802) Part of sip.conf: [laptop] type=friend ;dtmfmode=rfc2833 ;

[Asterisk-Users] DTMF

I am trying to connect to a vocal server from an asterisk server. A call is received via iax2 to my asterisk server. I then initiate a SIP connection to the vocal server. everything works great except dtmf doesnt work. A cisco 5300 can connect to this vocal server and do dtmf without a

Re: [Asterisk-Users] wireless

[EMAIL PROTECTED] wrote: Has anyone got a mobile wireless phone working with * yet Is it possible to use the Cisco 7920 with skinny Not sure, send me one and I'll test it for you. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL

Re: [Asterisk-Users] wireless

AFAIK the 7920 needs CallManager to work - if you haven't got that you'll have to wait for Cisco to make a general purpose version - or maybe buy a Pulver phone http://www.pulverinnovations.com/ - assuming that works with * Iain --On Monday, November 17, 2003 6:31 am -0500 Jeremy McNamara

[Asterisk-Users] voicepulse working?

hi, i am having trouble connecting using voicepulse. is it down? anybody has been using it today? thankx cm = Designs __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree

Re: [Asterisk-Users] voicepulse working?

- Original Message - From: C M [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 17, 2003 11:09 PM Subject: [Asterisk-Users] voicepulse working? hi, i am having trouble connecting using voicepulse. is it down? anybody has been using it today? I've used it quite a bit

[Asterisk-Users] IAX2 connectivity problem (qualify=yes)

Hi there, I still have issues with the IAX connection between two servers (one static (server A), one dynamic (server B), none behind NAT): B registers with A, and iax2 show registry shows that everything is fine. However, after a while if I check on server A with iax2 show peers I see a

[Asterisk-Users] Voip providers U.S (eastern) ??

Hello, can anyone out there recommend any voip companies in the US that can provide D.I.D lines that ring in to you SIP/IAX connection? I am aware of getting a T1 hooked up with D.I.D lines and a bank ofr lets say 20 numbers.. But is there any way around this using mostly VOIP ?? Thanks...

Re: [Asterisk-Users] Voip providers U.S (eastern) ??

- Original Message - From: jaycard [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 17, 2003 11:38 PM Subject: [Asterisk-Users] Voip providers U.S (eastern) ?? Hello, can anyone out there recommend any voip companies in the US that can provide D.I.D lines that ring in

Re: [Asterisk-Users] ISDN debugging and SIP dial-in issue]

Try this - change exten = s,3,Background(transfer); schaefer to exten = s,3,Playback(transfer); schaefer and then dial 3 from your GS. You should also add to sip.conf for [17476691152]: disallow=all allow=ulaw allow=alaw Are you sure you need the dtmfmode=inband for the GS? I don't

Re: [Asterisk-Users] IAX2 connectivity problem (qualify=yes)

Philipp von Klitzing wrote: Hi there, I still have issues with the IAX connection between two servers (one static (server A), one dynamic (server B), none behind NAT): B registers with A, and iax2 show registry shows that everything is fine. However, after a while if I check on server A with

[Asterisk-Users] Meetme : Zaptel ztdummy errors

Hello All !!! I trying to make meetme working! I don't have zaptel interface and I cannot install ztdummy cause I don't have usb-uhci !!! Is there any way to get it working ??? Thanks in advance, Areski # modprobe ztdummy /lib/modules/2.4.18-18.7.x/kernel/drivers/usb/usb-uhci.o:

RE: [Asterisk-Users] FXO Cards in Australia

On Mon, 2003-11-17 at 23:53, Adam Goryachev wrote: Yes, echo problems do still exist, I would suggest testing it before going live. Yeah, so I've heard. A couple of points to note: 1) Using soft phones seems to compound the issue So the echo problems are not so bad when using software

Re: [Asterisk-Users] IAX2 connectivity problem (qualify=yes)

Hi! Try qualify=3000 which will increase the time between checks.. Although it sounds like there is more to this problem than just increasing the time.. That's not really what I want to do - quality is really bad if you go above 2000, so it makes sense to keep it at this. I can schedule a

RE: [Asterisk-Users] FXO Cards in Australia

Re: these problems with the NetJet Cards: have people spoken with Traverse about them? I have found them to be most helpful with any problems (mainly with the Pulsar PCI ADSL cards) Try talking to [EMAIL PROTECTED] ? -Bryan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL

RE: [Asterisk-Users] FXO Cards in Australia

I'll be speaking to Guy tomorrow about this. Guy is certainly a helpful friendly guy and I'm sure he'll be keen to hear about these echo problems. Regards, Gonzalo On Tue, 2003-11-18 at 00:48, Bryan Nolen wrote: Re: these problems with the NetJet Cards: have people spoken with Traverse about

Re: [Asterisk-Users] Meetme : Zaptel ztdummy errors

Does your motherboard have USB ports and it's just that it's not loaded the right driver? Send us a: cat /proc/interrupts - Andrew Thompson - Original Message - From: Areski [EMAIL PROTECTED] To: Asterisk-Users Mailing-list [EMAIL PROTECTED] Sent: Monday, November 17, 2003 8:09

[Asterisk-Users] SIP soft phone registration

Hello-- I've installed a few X-Lite softphones on windows machines, and am playing with the settings. I've written before about this, and since have discovered that the X-Lite has an option to turn off registration, which I have set, because I bumped into a letter from Mark, saying that

Re: [Asterisk-Users] DTMF

I think that you are thinking of SIP INFO messages if you are expecting to see something in the SIP messaging. RFC2833 is sent as part of the RTP packets so you are not going to see a plain text 1,2,3,4,etc in a trace when using it. http://www.faqs.org/rfcs/rfc2833.html Sean - Original

Re: [Asterisk-Users] NuFone International Calls

On Sunday 16 November 2003 12:52 pm, [EMAIL PROTECTED] wrote: Hello Jasim. lag just took on a whole new definition :) -- Old mail I stored for the future for when I have time to look at asterisk again. Like now :-) Basically, I'm going over the scripts/files and resetting and testing.

Re: [Asterisk-Users] Meetme : Zaptel ztdummy errors

HI guys I do have usb-uhci. How do I build ztdummy? I think once its built I just have to do a modprobe to load it, I just don't know how to load it. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] iconnecthere incoming

Hi guys I just registered an incoming number with iconnecthere and I'm trying to set up incoming calls from icconnecthere on my asterisk server. I took a look at john todds sample sip.conf and extensions.conf file but for some reason my incoming is still not working. At this point I wish to

[Asterisk-Users] Meetme : Zaptel ztdummy errors

Hi again ! I don't know if the motherboard have usb port, the machine is a remote computer ;( Actually, I can install usb-ohci but not usb-uhci... cat /proc/interrupts CPU0 0: 2241860601 XT-PIC timer 1: 4 XT-PIC keyboard 2: 0

Re: [Asterisk-Users] Message Waiting

Page through this thread... http://lists.digium.com/pipermail/asterisk-users/2003-November/027220.html This was discussed just a few days ago. - Andrew Thompson - Original Message - From: Thorsten Neumann [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 17, 2003

Re: [Asterisk-Users] Transfer directly to voicemail?

You can do something like: exten = _8,1,Voicemail(${EXTEN:1}) This assumes 4 digit extesnions () and takes you to the voicemail box of whenever anyone dials 8. After you have this in your dialplan, you can transfer to it as easily as you can call it. Sean - Original

RE: [Asterisk-Users] Transfer directly to voicemail?

What I have done is created a separate set of extensions for just this scenario. I don't have my * box on the network right now (long story, and not very interesting), but the configs were pretty easy. I have extension X be my phone, and X+1 be a direct voicemail extension. All odd extensions

Re: [Asterisk-Users] Transfer directly to voicemail?

[EMAIL PROTECTED] wrote: Does anyone have a setup (or am I missing a simple thing here) to transfer a caller directly to someone voicemail? Example: I receive a call, the caller wants to speak with x, who I know is not in the office. Other than transferring them to x's extension, which rings to

[Asterisk-Users] GSM or WAV files for musiconhold?

Hello, Is there any way to use GSM or WAV files for the MusicOnHold application? All I've seen so far is that mpg123 is the only way to use the musiconhold application. It would be nice to not have the extra processor drain as well as the extra mpg123 install to worry about to get Moh

Re: [Asterisk-Users] mpg123 causing Asterisk Freeze?

mattf wrote: Hello, I am currently using MusicOnHold(mpg123), and it works just fine, but every once in a while I will get a flurry of warnings in the CLI like those below and Asterisk will freeze completely, and the only way to come out of it is with a kill -9 . Is mpg123 causing my problem? Is

[Asterisk-Users] IAX2 and MWI

Hi all, There is any MWI implementation in IAX2 to be used by a software IAX2 phone? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Can asterisk do this?

Here's what I want to make it do, have 2 servers acting as call agents handling inbound and outbound calls to a seperate pstn gateway(actually 2 of them but that is a seperate issue), and 2 servers acting as sip proxies for ata186's spread throughout several offices all remote. The call volume

[Asterisk-Users] mpg123 core when stopping asterisk

I typically start asterisk with the safe_asterisk script: 22865 pts/3S 0:00 /bin/sh /usr/sbin/safe_asterisk 22867 pts/3S 0:31 asterisk -vvvg -c 22871 pts/3S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 For-You.mp3 for.m 22873 pts/3S 0:00 mpg123 -q -s --mono -r 8000

[Asterisk-Users] Transfer directly to voicemail - Solved

Sean and WipeOut...that's greatexactly what I'm after. Works great. Daryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] problems with alsa (card ac97) in asterisk

Hello, I have asterisk 0.5.0, asterisk-oh323-0.5.6, openh323-1.12.2 and pwlib 1.5.2 compiled and installed. I have modules alsa 0.9.8 compiled and installed My PC have and audio card ac97 chipset intel i810 in motherboard. The list of the modules loaded is: namor:/etc/asterisk# lsmod Module    

Re: [Asterisk-Users] VOIP phonesets vs. cheap Analog touch-tone sets with Asterisk

Steve Murphy wrote: Hello-- I've been asked an interesting question, and I'm too ignorant to answer it authoritatively (yet). Can anyone help me? Question: If I'm going to implement a somewhat small (10-80) phone system, and I have a choice of using VOIP phoneset (like SNOM or Grandstream or

RE: [Asterisk-Users] mpg123 causing Asterisk Freeze?

Hello, Well, I've re-encoded all of my files with lame just like it says to for Constant Bitrate: prompt$ lame -b 128 -F file1.wav file1.mp3 but I'm still getting the error at Asterisk start time: [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf':

Re: [Asterisk-Users] IAX2 and MWI

yes, it's provided as an IE during registration. Mark On Mon, 17 Nov 2003, Dan wrote: Hi all, There is any MWI implementation in IAX2 to be used by a software IAX2 phone? Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED]

Re: [Asterisk-Users] voicepulse working?

On Mon, Nov 17, 2003 at 04:09:20AM -0800, C M wrote: hi, i am having trouble connecting using voicepulse. is it down? anybody has been using it today? thankx cm Just did a test call. Went through fine. Jayson ___ Asterisk-Users mailing list

Re: [Asterisk-Users] mpg123 core when stopping asterisk

Hi. also here happens the same thing. btw, if you don't need asterisk to dump cores in case of crash, just modify safe_asterisk, where it has ulimit -c unlimited to ulimit -c 0 . This prevents asterisk for dumping and also affects related processes. Matteo Scrive Rich Adamson [EMAIL PROTECTED]:

[Asterisk-Users] Sample proposal

Hi: I have a client who would like a proposal for a small Asterisk system. Does anyone know of a boilerplate proposal that I can copy and modify? This will include hardware and setup of Asterisk. Thanks Costas -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] --

RE: [Asterisk-Users] Updated Asterisk-NL

-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Cees de Groot Sent: maandag 17 november 2003 8:57 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Updated Asterisk-NL I have updated the voice prompts (mostly small fixes, but still work left to be

[Asterisk-Users] qtelnet product, FXO gateway

I just received this email in response to a question I asked of Qtelnet about their hardware's compatibility. I had specifically asked about their FreeRide FXO-1301 Gateway. Now, if someone would be so kind as to send me a demo unit, I'd be glad to test it :-) - Andrew Thompson -

Re: [Asterisk-Users] DTMF

I dont expect to see an ascii code or such since the tones are in a rtp stream. But when I place the dtmf type to "info" in the sip.conf and make a call I see this under asterisk with sip debug on. DEBUG[122896]: File rtp.c, Line 942 (ast_rtp_raw_write): Difference is 960, ms i s 140 I

RE: [Asterisk-Users] Your thoughts..

We have a cron job that moves master.csv to subdir/timestamp.csv and then tries to parse and delete all csvs in that directory. The reasons we did this were 1) We wanted to store more data in the database without having to change source code. For example, we have 45 asterisk servers writing to

Re: [Asterisk-Users] strange Music on Hold between SNOM, Grandstream and Asterisk

I have the exact problem with a little different configuration. I am using GS phones and a sip/fxo gateway. I make a call from PSTN == sip/fxo gateway == * == sip phone (GS). Life is good until GS places call on hold. What I see on the wire: 1 - RTP packets every 20msec both directions for a

[Asterisk-Users] Static Config?

Hi, I'm new to asterisk. After fiddling a bit I got it to work. It seems great. One question though, is it possible to configure asterisk when it is running? i.e. add new phones or do you have to restart it every time you want to make changes? Thanks, Storm.

RE: [Asterisk-Users] Putting call on hold

Want to put the caller on hold and them hear music by pressing the hold button -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Posted At: Friday, November 07, 2003 9:56 PM Posted To: Asterisk User Group Conversation:

[Asterisk-Users] Re: Updated Asterisk-NL

Arnold Ligtvoet [EMAIL PROTECTED] said: how do I use the patch included in the tar.gz? I have downloaded the * source from cvs, so the * source in in /usr/src/asterisk/ ? err... 'man patch'? ;-) % cd /usr/src/asterisk; patch -p0 /path/to/asterisk-nl-patch should work in most circumstances.

[Asterisk-Users] Wifi600 problem

Some of you have got Wifi600 wireless SIP phone working with Asterisk. Specially John Todd ( nice review ). My phones register ok. They can also receive calls from other phones. But for some reason I can't make them call out ( anybody, ie. SIP or PSTN ). This seems to be due to the phone not

Re: [Asterisk-Users] Static Config?

On Mon, 2003-11-17 at 12:37, Storm D. J. Petersen wrote: Hi, I'm new to asterisk. After fiddling a bit I got it to work. It seems great. One question though, is it possible to configure asterisk when it is running? i.e. add new phones or do you have to restart it every time you want to

[Asterisk-Users] help voicepulse connect

Hi All I signed up for an account with voicepulse connect service and received the info to set up asterisk. Anyonehave that confs to send as an example? Thanks Miklos

Re: [Asterisk-Users] ISDN debugging and SIP dial-in issue]

On Mon, Nov 17, 2003 at 02:08:30PM +0100, Philipp von Klitzing wrote: You should also add to sip.conf for [17476691152]: disallow=all allow=ulaw allow=alaw This was the key. I now hear the voice prompts correctly. rather use the new syntax for the Dial application like

Re: [Asterisk-Users] DTMF

I apologize. When you said that you were looking in the SIP debug, I thought that you were expecting the rfc2833 to be a SIP message. You might could do something with Ethereal that would show you what is going on. Sean - Original Message - From: Scott England To:

Re: [Asterisk-Users] zap show channels - No such command....

Got it to work. Thanks. Sri wrote: From the listserv, i found this command to see the zap channels setup. but when i execute it, i get no such command... is there any other command i can use to see the configured channels.. (not active channels...) *CLIzap show channels No such command 'zap'

RE: [Asterisk-Users] Static Config?

One more important thing to note: you can issue a restart when convenient (or type resttabwtabtab if you're lazy like me). This waits until the channels are free (correct me if I'm wrong). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf

[Asterisk-Users] Hunt groups and SIP?

I would like to setup a hunt group, not a group ring, using sip phones. Anyone done this with sip devices? Comments suggestions? I have not had much luck with the outgoinglimit=1, incominglimit=1 stuff that I would need to get busy extinctions to work right, which is why I'm asking on the list.

Re: [Asterisk-Users] Re: Updated Asterisk-NL

Hi, Citeren Cees de Groot [EMAIL PROTECTED]: Arnold Ligtvoet [EMAIL PROTECTED] said: how do I use the patch included in the tar.gz? I have downloaded the * source from cvs, so the * source in in /usr/src/asterisk/ ? err... 'man patch'? ;-) % cd /usr/src/asterisk; patch -p0

[Asterisk-Users] unsubscribe

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RE: [Asterisk-Users] Hunt groups and SIP?

Look at the Queue application... it will probably fulfill your needs. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Sizemore Sent: Monday, November 17, 2003 2:25 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Hunt groups

[Asterisk-Users] Radius on *

Does Asterisk support Radius accounting? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: Lunes, 17 de Noviembre de 2003 12:08 p.m. Para: [EMAIL PROTECTED] Asunto: Asterisk-Users digest, Vol 1 #1912 - 11 msgs Send

Re: [Asterisk-Users] Hunt groups and SIP?

- Original Message - From: James Sizemore [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 17, 2003 3:24 PM Subject: [Asterisk-Users] Hunt groups and SIP? I would like to setup a hunt group, not a group ring, using sip phones. Anyone done this with sip devices? Comments

Re: [Asterisk-Users] problems with alsa (card ac97) in asterisk

Antonio Angel wrote: [chan_alsa.so] = (ALSA Console Channel Driver) asterisk: pcm.c:5486: snd_pcm_sw_params_set_silence_threshold: Assertion `val pcm-buffer_size' failed.Aborted There is problems between asterisk and alsa or ac97 ? I need any option for to compiler asterisk ? I guess most

Re: [Asterisk-Users] unsubscribe

Shoval, Please read the footer appended by the mailing list server on the end of every message. It points you to the proper location to unsubscribe. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED]

Re: [Asterisk-Users] DTMF

I can't resist asking. What do you think of Vocal as compared to *? Anything Vocal has but missing in *? -- Original Message -- From: Scott England [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Mon, 17 Nov 2003 02:58:55 -0800 I am trying to connect

[Asterisk-Users] New FAQ on Echo Cancellation

With the assistance of writers Stephen and Rich, I've edited two new pages on the Wiki. It's general information on Echo cancellation - how to attack it and locate it. You'll find them from the FAQ: http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ Still looking for more advice, tips and

[Asterisk-Users] SIP calls no longer work

Hello, I'm having a problem with SIP. More specifically, I can't make any calls using SIP. I have had an iConnectHere account and Free World Dialup account working for quite some time, and now all of a sudden I can't make any SIP outgoing calls. PBX*CLI sip show registry Host

Re: [Asterisk-Users] Hunt groups and SIP?

Also, check my patch http://bugs.digium.com/bug_view_page.php?bug_id=408 which does fix incominglimit/outgoinglimit. Stops callwaiting on sip phones. Hope that helps. Paul - Original Message - From: David Gomillion [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, November 18,

Re: [Asterisk-Users] IAX2 connectivity problem (qualify=yes)

WipeOut wrote: Philipp von Klitzing wrote: Hi there, I still have issues with the IAX connection between two servers (one static (server A), one dynamic (server B), none behind NAT): B registers with A, and iax2 show registry shows that everything is fine. However, after a while if I check

Re: [Asterisk-Users] Meetme : Zaptel ztdummy errors

Areski wrote: Hello All !!! I trying to make meetme working! I don't have zaptel interface and I cannot install ztdummy cause I don't have usb-uhci !!! Is there any way to get it working ??? FAQ http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ See the page on Zaptel timers, called

Re: [Asterisk-Users] Sample proposal

If there are not replies on the list could you please forward any replies to such a proposal? I too have a client who wants me to develop a system. I just wish documentation for Asterisk was better overall. Thanks! -Andrew On Monday 17 November 2003 09:58, costas wrote: Hi: I have a

Re: [Asterisk-Users] Your thoughts..

David Carr wrote: We have a cron job that moves master.csv to subdir/timestamp.csv and then tries to parse and delete all csvs in that directory. The reasons we did this were 1) We wanted to store more data in the database without having to change source code. For example, we have 45 asterisk

Re: [Asterisk-Users] unsubscribe

How dare the answer be right in front of them :-D I'm sorry, I just had to get in on it once... I promise, I won't do it again... - Andrew Thompson - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 17, 2003 3:43 PM Subject:

[Asterisk-Users] Asterisk Cdr

Hi All, I am a new user of Asterisk. I have compiled and instaled asterisk-0.5.0. I have also added a set of new dial plans along with some new users in my extension.conf file.I have also edited sip.conf file.My intention was,to run asterisk as a IVR and billing server.I am also using VOCAL

RE: [Asterisk-Users] wireless

Don't sound bad do they Except their ship date is mid January 2004 Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson Sent: Monday, 17 November 2003 10:34 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] wireless

Re: [Asterisk-Users] Radius on *

Sebastian Nocetti wrote: Does Asterisk support Radius accounting? No and there is absolutely no need for it to. RADIUS is not anything that should have ever been deployed in a VoIP environment. There are many methods to talk directly to a database, why add another layer of

Re: [Asterisk-Users] IAX2 connectivity problem (qualify=yes)

- Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 17, 2003 3:58 PM Subject: Re: [Asterisk-Users] IAX2 connectivity problem (qualify=yes) WipeOut wrote: Philipp von Klitzing wrote: Hi there, I still have issues with

Re: [Asterisk-Users] VOIP phonesets vs. cheap Analog touch-tone sets with Asterisk

bottom response = on On Mon, 2003-11-17 at 08:34, Steve Murphy wrote: Hello-- I've been asked an interesting question, and I'm too ignorant to answer it authoritatively (yet). Can anyone help me? Question: If I'm going to implement a somewhat small (10-80) phone system, and I have a

Re: [Asterisk-Users] SIP calls no longer work

- Original Message - From: jerk face [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 17, 2003 3:53 PM Subject: [Asterisk-Users] SIP calls no longer work Hello, I'm having a problem with SIP. More specifically, I can't make any calls using SIP. I have had an

[Asterisk-Users] 3Com NBX phones

Has there been any luck getting the 3Com NBX series phones to work with Asterisk? Thanks! -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] asterisk and Codec G-723

Hello I want that all the information of voice that travels by asterisk works with codec g-723 This can be done? Javier Rios

Re: [Asterisk-Users] DTMF

Vocal is a complex system and its very Cisco centric (since Cisco funds it I'm not suprised) but in all its been good. We went with it for the SIP support and the fact that our version is heavily modified to intgrate with our custom app. If * had solid SIP support 2 years ago things might have

Re: [Asterisk-Users] Distintive Ring on x100p

Fantastic, works brilliantly, this should be in the CVS source, it would be good if the MSN could be set and show up in the cdr? Robb Brian West wrote: 1 dont start in debug mode.. you will get lost in all the garbage start with safe_asterisk then asterisk -r set verbose 4 call in to each number

Re: [Asterisk-Users] unsubscribe

On Mon, 2003-11-17 at 15:19, Andrew Thompson wrote: How dare the answer be right in front of them :-D I'm sorry, I just had to get in on it once... I promise, I won't do it again... This points out another really bad thing about HTML email. At least some of the mail clients are sending proper

Re: [Asterisk-Users] Radius on *

Dude, you deserve a double dipping in flaming hot sauce for this stupid post. I'll give you that the question isn't stupid, just the way you went about asking it. First you point out why I hate people who use digest format of a mailing list that is many to many communications. You just sent the

Re: [Asterisk-Users] Radius on *

Does Asterisk support Radius accounting? No and there is absolutely no need for it to. RADIUS is not anything that should have ever been deployed in a VoIP environment. There are many methods to talk directly to a database, why add another layer of complexity and point of failure?

Re: [Asterisk-Users] MP3Player problem -repost

On 17 Nov 2003 13:39:20 +0100, Areski [EMAIL PROTECTED] wrote: I tried also to enter directly this instruction: mpg123 -w ki.wav http://digitaljukebox.com/Carta.mp3 And I get : HTTP request failed: HTML PUBLIC -//IETF//DTD HTML 2.0//EN The file exist, I get do a wget on it... Some ideas how to get

Re: [Asterisk-Users] asterisk and Codec G-723

On Mon, 2003-11-17 at 16:06, Javier Rios wrote: Hello I want that all the information of voice that travels by asterisk works with codec g-723 This can be done? Need more information. You mention G723 so it is assumed you are talking VoIP, but there are at least 2 widely accepted

Re: [Asterisk-Users] asterisk and Codec G-723

On Monday 17 November 2003 16:06, Javier Rios wrote: Hello I want that all the information of voice that travels by asterisk works with codec g-723 This can be done? No. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] RE: Asterisk-Users digest, Vol 1 #1918 - 9 msgs

An example for Radius is calling cards.. I can use * for this kind of service... With platforms that use Radius Server. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: Lunes, 17 de Noviembre de 2003 07:16 p.m. Para: [EMAIL

Re: [Asterisk-Users] 3Com NBX phones

3com made a phone that supported SIP but was discontinued. From what I have heard, the hardware was exactly the same, just a different BIOS. I would like to try or hear of someone trying to flash an NBX phone into a SIP phone. Here is a link to one that sold on Ebay.

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