Re: [asterisk-users] asterisk audits

On 9/27/07, Tilghman Lesher <[EMAIL PROTECTED]> wrote: > > On Wednesday 26 September 2007 18:39:31 Mark Quitoriano wrote: > > Some company asked me to do audits with there asterisk boxes. Is there a > > standard that i should be following in auditing? anyway can give me a > start > > what to do wit

Re: [asterisk-users] How to "busy out" zap channels

Brian: Maybe the CLI command "stop gracefully" is what are you looking for. Basically, Asterisk will stop receiving incoming calls (of any channel type) and stop itself when all the current calls finish. I hope this help you. Best regards, Tomás. 2007/9/26, Brian Roy <[EMAIL PROTECTED]>: > I know

Re: [asterisk-users] asterisk audits

On Wednesday 26 September 2007 18:39:31 Mark Quitoriano wrote: > Some company asked me to do audits with there asterisk boxes. Is there a > standard that i should be following in auditing? anyway can give me a start > what to do with asterisk audits? Have you considered the ethics of getting yours

Re: [asterisk-users] Asterisk realtime error

Peder, I have all the permissions in mysql user. I can query my database from the local box. Mik Cheez, yes, it is. mysql.sock is in /var/lib/mysql/ Asterisk and Mysql are in the same PC I still have the same error and don't know what to do. help plz! thanks in advance, Renzzo Mik Cheez wrote:

[asterisk-users] help with channelbank audiocodes MP-124

Hi: We're offering some sort of reward to that one who can help us For this site we are using trixbox and Asterisk 1.2 More info off list. Many thanks, Carlos ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwid

Re: [asterisk-users] Asterisk - Spandsp Fax not working?

Jonn R Taylor wrote: > marco britannio wrote: >> Hi all, >> I'm trying to setup an asterisk based fax receiving machine. >> i'm using asterisk 1.2.18 and app_rxfax with spandsp 0.0.4pre9 >> I have no problems with a modem-fax, but with the fax machines i have >> tried almost every fax fails, both

Re: [asterisk-users] Asterisk - Spandsp Fax not working?

marco britannio wrote: > Hi all, > I'm trying to setup an asterisk based fax receiving machine. > i'm using asterisk 1.2.18 and app_rxfax with spandsp 0.0.4pre9 > I have no problems with a modem-fax, but with the fax machines i have > tried almost every fax fails, both in sending and receive. > th

[asterisk-users] Doesn't seem to want to transcode.

Hi all, I've got a .wav file on my asterisk server and I've got an extension that plays it back. When I dial the extension on the local server, it plays back just fine. When I create a call file that calls a (remote) pstn phone number and plays that file, it works just fine, also. But, when

[asterisk-users] voip hacking article

http://www.informationweek.com/news/showArticle.jhtml?articleID=20210178 1 Nothing deep and meaningful in the article but worth a read. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial).

Re: [asterisk-users] ChanSpy issue

You are technically correct, its just a shorthand. on Wednesday 09/26/2007 "Eric \"ManxPower\" Wieling"([EMAIL PROTECTED]) wrote > There is no such thing as a "SIP Trunk" in Asterisk. Nope. It does not > exist. Some people (seems to me mostly GUI people) use the term "SIP > trunk" to mean

Re: [asterisk-users] ChanSpy issue

There is no such thing as a "SIP Trunk" in Asterisk. Nope. It does not exist. Some people (seems to me mostly GUI people) use the term "SIP trunk" to mean "SIP friend/user/peer". John covici wrote: > I am not an expert on chanspy, but it seems to me spying on the trunk > would not work very w

[asterisk-users] h.323 out of media path

Hi folks !!! Is there a way to have asterisk out of the media path, when using H.323 ? I mean, it would be better to have something like sip's REINVITE... is that possible? Thanks in advance... -lars ___ Sign up now for AstriCon 2007! September

Re: [asterisk-users] ChanSpy issue

I am not an expert on chanspy, but it seems to me spying on the trunk would not work very well, would not you hear multiple conversations mixed if more than one extension were calling? Seems best to me to spy on an extension. YOu also can do a show channels to see who is talking to whom. on Wedn

Re: [asterisk-users] IAX gsm bandwith calls

If you've got a bandwidth of something that low you'll probably want to use g723.1 or g729 on this line. If your lucky you'll be able to place two calls at once over this link. You won't be able to do anything else though. Tom _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On B

Re: [asterisk-users] Asterisk realtime error

Is your mysql.sock actually in /var/lib/mysql/ ? RENZZO SOTOMAYOR wrote: > Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using > Idefisk softphones. I followed the steps of "how to" of voip-org but > always have this error: > > Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c

Re: [asterisk-users] Asterisk realtime error

Could be a mysql permission issue. Try this from the local box: mysql -u root -p use asterisk; select * from sip_buddies; select * from iax_buddies; If you get that far and can see the entries in iax_buddies and sip_buddies, you know it isn't a permissions issue. If you can't, then you know

[asterisk-users] asterisk audits

Hi, Some company asked me to do audits with there asterisk boxes. Is there a standard that i should be following in auditing? anyway can give me a start what to do with asterisk audits? thanks! -- Regards, Mark Quitoriano, CCNA Fan the flame... http://www.spreadfirefox.com/?q=user/register&r=1

[asterisk-users] Asterisk realtime error

Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk softphones. I followed the steps of "how to" of voip-org but always have this error: Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info

Re: [asterisk-users] Music On Hold

Concatenate the files into one larger file, in the order you want them to play > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Joel Hill > Sent: Wednesday, September 26, 2007 7:01 PM > To: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] Networking Question

A few questions for you: Where is your DNS Server for your LAN located by using the 172.17.x.x address I suppose there is more to your network than two segments, (Asterisk may drop connections if it has a problem with DNS) How are your Polycom phones configured? Are they using a ftp/tftp se

Re: [asterisk-users] Music On Hold

> > Hi All, > > > > I need to have the same file played from MoH every time someone gets > > to > > MoH from a Dial. I want to play marketing messages from it and I > > want it > > to start from file 1 every time. > > > > Anyone know if/how this can be done? > On Wed, 2007-09-26 at 09:36 -0400, Fo

Re: [asterisk-users] HOWTO/FAQ question (Location: Sweden)

Yes you can :) that's what asterisk can do. Im running all sip in my asterisk in my 2 call centers. that all "SIP" On 9/26/07, Turbo Fredriksson <[EMAIL PROTECTED]> wrote: > > > "zoachien" == zoachien <[EMAIL PROTECTED]> writes: > > zoachien> Turbo Fredriksson wrote: > >> How do I con

Re: [asterisk-users] Music On Hold

Thanks for the suggestion, but I need it to play multiple messages. Always starting with the same one. Cheers, Joel. On Wed, 2007-09-26 at 09:36 -0400, Forrest Beck wrote: > Make the file the only one in the /var/lib/asterisk/moh directory. > > Forrest Beck > [EMAIL PROTECTED] > www.shift8.biz

Re: [asterisk-users] ChanSpy issue

The parameter to Chanspy should be the whole or part of the channel name. I do not understand what you mean by "sip trunk". It make perfect sense that you can hear both streams of voice when you use the phone's extension as Asterisk usually uses "SIP/extension+xxx" as the channel name of the cal

Re: [asterisk-users] Ast_log

Thanks to all who replied. -Original Message- From: [EMAIL PROTECTED] on behalf of Ed Nuñez Sent: Wed 9/26/2007 4:28 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Ast_log The Asterisk log file is normally located in /var/log/asterisk Bu

Re: [asterisk-users] ChanSpy issue

Hello list I am having an issue with Chanspy/SIP that I’m hoping someone has come across and resolved in the past. I am sending calls that come in TDM through T1 ZAP channels and go out to a SIP trunk. If I spy on the SIP channel, I can hear the person on the SIP side of the call jus

Re: [asterisk-users] Ast_log

The Asterisk log file is normally located in /var/log/asterisk But you may want to read your asterisk.conf file to make sure the path in which your system store it. You will see something like this astlogdir => /var/log/asterisk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMA

Re: [asterisk-users] Networking Question

Do your phones have the 172.17.x.x as the proxy address? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian M. Arlinghaus Sent: Wednesday, September 26, 2007 4:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-u

Re: [asterisk-users] Ast_log

Wai Wu wrote: > Hi all, > Anyone know where the asterisk log file is stored? I have some failed > /var/log/asterisk Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." _

[asterisk-users] Networking Question

I have an Asterisk server running REL 4 with two NICs. One NIC has a 192.168.1.x IP address and is connected to a POE switch with Polycom phones that have 192.168.1.x IP addresses. The other NIC has a 172.17.x.x IP address connected to a router. The router is connected to the Internet. If the

Re: [asterisk-users] How to "busy out" zap channels

Very nasty indeed. Through my experience with PRI, the TelCo switchs are not that present to deal with. Your method will work, kind of. However, if the TelCo decides to send you a call during that split second of idle, how are you going to handle it. The best way is still to call your TelCo to take

[asterisk-users] How to "busy out" zap channels

I know this topic came up many months back and some discussions were being had on how to do this within the Zaptel drivers. However, I'm looking for even a crude hack that someone has put together to get this done. We have PRI's and LD T1's that are load balanced on two boxes. The hunt order goes

[asterisk-users] Ast_log

Hi all, Anyone know where the asterisk log file is stored? I have some failed calls into my Asterisk box, and I just want to find out why those calls failed. Thnx. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Ban

Re: [asterisk-users] Music On Hold

So concatenate all the files you've got into one to follow Forrest's suggestion :) Forrest Beck wrote: > Make the file the only one in the /var/lib/asterisk/moh directory. > > Forrest Beck > [EMAIL PROTECTED] > www.shift8.biz > > > > On Sep 26, 2007, at 3:07 AM, Joel H

[asterisk-users] Slightly OT: Help choosing a free software license?

I'm a little boggled by all the license models one can choose to release a piece of software under. GPL, AFPL, etc? I'm hoping someone can point me to a CLEAR resource that talks about the pros and cons of choosing one over another. All I've found seems to go right over my head. (it's basica

Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?

yea thats what i did i put SIP 1.6 and its working like a champ, there should be a way to get it working with 2.2, i'll wait for my next 601 and play with it. On 9/26/07, Doug <[EMAIL PROTECTED]> wrote: > > At 00:18 9/26/2007, Al lists wrote: > >One more thing i noticed today, > >with SIP 2.2 and

Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?

At 00:18 9/26/2007, Al lists wrote: >One more thing i noticed today, >with SIP 2.2 and Polycom 601 i wasnt able to enable buddy watch to >use with hints. >I'll spend more time on it later to see what is up with that. I guess they still haven't fixed that. The 601 that we have is using: 1.6.7

Re: [asterisk-users] faster timeout in ENUMLOOKUP() function

I need to change my email address for this list but the website is having issues doing that. Can anyone give me another method to accomplish this? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho Sent: Wednesday, September 26, 2007 10:32 AM To: asterisk-users@l

Re: [asterisk-users] Yikes! Polycom 501 chokes on BootRom 4.0.0?

I use a 650, so YMMV, but it's working with mine. Mike _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al lists Sent: Wednesday, September 26, 2007 01:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Yikes! Polycom 501 chokes

[asterisk-users] IAX gsm bandwith calls

Hi everybody I have 2 asterisk server connected by iax trunk using gsm over a 64Kbps Frame relay circuit, my questions are:whats is bandwith of each call?, and how to limit this on asterisk? Thanks.. ___ Sign up now for AstriCon 2007! September 25-28th

[asterisk-users] Fwd: Routing issue

Hi list I'm kinda new to asterisk and I'm woriking for a company that sells Asterisk solutions and appliances. I installed TrixBox on a litle PC @ home and a x100p card which is recognized as a Zaptel card, I made some in/outbound routes and they seem to work but I have a problem with SIP softpho

[asterisk-users] Routing issue

Hi list I'm kinda new to asterisk and I'm woriking for a company that sells Asterisk solutions and appliances. I installed TrixBox on a litle PC @ home and a x100p card which is recognized as a Zaptel card, I made some in/outbound routes and they seem to work but I have a problem with SIP softpho

[asterisk-users] Asterisk - Spandsp Fax not working?

Hi all, I'm trying to setup an asterisk based fax receiving machine. i'm using asterisk 1.2.18 and app_rxfax with spandsp 0.0.4pre9 I have no problems with a modem-fax, but with the fax machines i have tried almost every fax fails, both in sending and receive. the machines are sending a receiving a

[asterisk-users] faster timeout in ENUMLOOKUP() function

Hi all, In my server dialplan, it first tries to dial possible SIP URI contacts returned by DNS lookups using the ENUMLOOKUP function; it only sends calls to PSTN when there aren't any NAPTR records of the dialed number. Problem arises when my Internet connection is down to some locations, which i

Re: [asterisk-users] Grandstream GXP2020 / 2000

On Sep 26, 2007, at 10:58 AM, Ricardo Carvalho wrote: > All phones have firmware version 1.1.1.14; we are testing new > stable version 1.1.4.18 but by now we found that some phones freeze > sometimes - version 1.1.1.14 seems more stable. I'm not sure which firmware I'm running on my GXP2000 (

Re: [asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME

Thxs!! On Wed, 2007-09-26 at 10:26 -0400, Doug Lytle wrote: > Luis Morales wrote: > > That's an good tips. Where i find information or help to provisioning > > > > http://www.voip-info.org/wiki-Polycom+Phones > > Doug > ___ Sign up now for Ast

Re: [asterisk-users] SIP Panel?

Yes, I have. It is not difficult. I use the Asterisk Manager interface. Is there a particular question? - Walt Terry Giufre-Sweetser wrote: > Dear List, > > Has anyone found or written a status panel application, windows or > linux, that uses SIP notifies and subscriptions, to gather the status

[asterisk-users] Manager Originate Action and Cancel

I'm using the Originate Action on the Asterisk Manager to place calls between two extensions in async mode. Is there any way to cancel the Originate Action before I get the OriginateResponse action? I'm unable to perform a Hangup because I can't know the channel name before I get the response...

Re: [asterisk-users] Grandstream GXP2020 / 2000

We've a site with about 200 Grandstream GXP2000 phones, and they work quite well. We made some CGI Perl scripts to mass-deploy and manage their configurations from a MySQL DB into a TFTP server, where the phones go to download their binaries. With some initial work, now it has become easy to manage

Re: [asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME

Luis Morales wrote: > That's an good tips. Where i find information or help to provisioning > http://www.voip-info.org/wiki-Polycom+Phones Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

Re: [asterisk-users] Grandstream GXP2020 / 2000

Erik Wartusch wrote: > Hi, > > Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a > business graded installation (with really traffic on not 3 calls a > day ;-) ) > Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall) > > Hi Erik, we have about 75 Grands

Re: [asterisk-users] Supermicro PDSME+ and TE110P [ ref:00D36mPe.50033qy57:ref ] NEW CASE 22828

On Wednesday 26 September 2007 08:36:01 kido wrote: > Mainboard: SUPERMICRO PDSME+ For whatever reason, I've seen a lot of issues with SuperMicro boards, which is why the reseller I've worked for tends to use Abit motherboards, not SuperMicro, as the Abit boards do not exhibit these problems. --

Re: [asterisk-users] Zaptel-1.4.5.1 Compile Error

On Wednesday 26 September 2007 05:13:31 Jeng Yu wrote: > Thank you, Tilghman. Your suggestion did it. I ran > into similar compile problem later: > - > /usr/src/zaptel-1.4.5.1/xpp/xbus-sysfs.c:135: error: > unknown field âhotplugâ specified in initializer > make[4]: *** > [/usr/src/zapt

Re: [asterisk-users] # to transfer calls

From the asterisk CLI do "show features" you'll find # is default for Blind transfer your entry below is commented out, ie has a ;in_front_of_it hope this helps Bails VoIP Newbie wrote: > features.conf has default settings as follows: > ; > ; Sample Parking configuration > ; > > [general] >

Re: [asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME

That's an good tips. Where i find information or help to provisioning the phones with ftp ? In my case the setup was made on each phone using polycom web interface. Regards, Luis Morales On Wed, 2007-09-26 at 09:23 -0400, Doug Lytle wrote: > Luis Morales wrote: > > Doug, > > > > Where is locat

Re: [asterisk-users] Zaptel-1.4.5.1 Compile Error

On Wed, Sep 26, 2007 at 11:13:31AM +0100, Jeng Yu wrote: > Thank you, Tilghman. Your suggestion did it. I ran > into similar compile problem later: > - > /usr/src/zaptel-1.4.5.1/xpp/xbus-sysfs.c:135: error: > unknown field âhotplugâ specified in initializer > make[4]: *** > [/usr/src/za

Re: [asterisk-users] chan_sip falls over with undefined symbolast_pickup_ext

Silly me. I solved it myself. I was not loading res_features.so Steve Hindmarch From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: 26 September 2007 14:31 To: asterisk-users@lists.digium.com

Re: [asterisk-users] Supermicro PDSME+ and TE110P [ ref:00D36mPe.50033qy57:ref ] NEW CASE 22828

Hello, Digium support kindly proposed to ship a TE120P card to help resolve the issue. I plugged in the card, and introduced the loopback plug. I cleared the red alarm for a while and then i started seeing alarm switching from Yel/Recovering to Blue/Rec with a lot of IRQ Misses. I call Digium

Re: [asterisk-users] Music On Hold

Make the file the only one in the /var/lib/asterisk/moh directory. Forrest Beck [EMAIL PROTECTED] www.shift8.biz On Sep 26, 2007, at 3:07 AM, Joel Hill wrote: Hi All, I need to have the same file played from MoH every time someone gets to MoH from a Dial. I want to play marketing

[asterisk-users] chan_sip falls over with undefined symbol ast_pickup_ext

I have just downloaded and built asterisk 1.4.11 on my Fedora Core 6 box. All seemed to go well but once I had configured the server for SIP and sent my first SIP call to the server then asterisk crashed with the message *CLI> asterisk: symbol lookup error: /usr/lib/asterisk/modules/chan_sip.so

[asterisk-users] My G729 problem re-visited

Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-th

Re: [asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME

Luis Morales wrote: > Doug, > > Where is located sip.cfg file ? > Where ever you are provisioning your phones from. I do my provisioning with FTP and the files are located in the polycom home directory that I created. Doug -- Ben Franklin quote: "Those who would give up Essential Lib

Re: [asterisk-users] # to transfer calls

VoIP Newbie wrote: > > features.conf has default settings as follows: > ; > ; Sample Parking configuration > ; I believe # is the default. If you don't define it, it will use that default. Set it to something that you know won't be used. Maybe ##3 Doug -- Ben Franklin quote: "Those who

Re: [asterisk-users] # to transfer calls

features.conf has default settings as follows: ; ; Sample Parking configuration ; [general] parkext => 700 ; What ext. to dial to park parkpos => 701-720 ; What extensions to park calls on context => parkedcalls ; Which context parked calls are in ;parkingtim

Re: [asterisk-users] Asterisk Redundancy

On 9/26/07, SIP <[EMAIL PROTECTED]> wrote: > > No. It's not. But there still exists the possibility even in a > relatively stable situation that the software could crash or that > hardware could fail. It's best, when planning a highly-available > solution, to plan for the unforeseen and not assume

Re: [asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME

Doug, Where is located sip.cfg file ? Regards, Luis Morales On Wed, 2007-09-26 at 08:32 -0400, Doug Lytle wrote: > Luis Morales wrote: > > Hi, > > > > Does any know adjust the volume for polycom ip soun point ? I adjust by > > the phone on the current call, but when hangup the volume lost the

[asterisk-users] Grandstream HT 502 ATA stops receiving calls

Hi, I have an annoying problem with the Grandstream HT 502 ATA. When the ATA first powers up, and registers to the asterisk server, everything works fine, and the ATA is able to receive and send calls, however after a period of time, the ATA can only make outgoing calls, but it is unable to rec

Re: [asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME

Luis Morales wrote: > Hi, > > Does any know adjust the volume for polycom ip soun point ? I adjust by > the phone on the current call, but when hangup the volume lost the > Look in your sip.cfg for the line: Change them from 0 to 1 Doug -- Ben Franklin quote: "Those who would give up E

[asterisk-users] configuration of wanpipe for asterisk.

Hi: I install A102 sangoma's card and connect E1 link it now for configuring wanpipe which one should I select for dial plan context:from pstn?or from internal? Best regards. - Be a better Globetrotter. Get better travel answers from someone who knows. Yah

[asterisk-users] POLYCOM - SOUND POINT IP 301 - VOLUME

Hi, Does any know adjust the volume for polycom ip soun point ? I adjust by the phone on the current call, but when hangup the volume lost the volume configuration. There are any way to set phone volume by default ? Regards, Luis Morales ___ Sign

Re: [asterisk-users] Asterisk Redundancy

Per Jessen wrote: > Atis Lezdins wrote: > > >> This seems nice way of sharing settings, however it wouldn't take over >> calls in progress. For us, currently the greatest problem is that >> whenever Asterisk crashes, calls are lost, and that means - lost >> money. Are there any ideas? >> >

Re: [asterisk-users] [on-asterisk] Configure one call per line on Cisco 7941/7961

Ahh. Differences with the 7961 software from that of the 7960's. Sorry, need to research more. - dbc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary T. Giesen Sent: September-26-07 12:29 AM To: David Cook Cc: [EMAIL PROTECTED]; asterisk-users@lists.d

Re: [asterisk-users] Busy problem

Erik Wartusch wrote: > - Got SIP response 486 "Busy Here" back from 172.10.3.31 > I see that response when someone presses the DND button on our Polycom phones. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither

[asterisk-users] Busy problem

Hi, I've a huge problem with the following: Setup: Asterisk 1.4.11 I've got two Thomson ST2030s in an queue. After a while Asterisk logs the following if somebody calls the queues number: - Got SIP response 486 "Busy Here" back from 172.10.3.31 -- SIP/office1-0823d190 is busy -- Nobo

Re: [asterisk-users] Zaptel-1.4.5.1 Compile Error

Thank you, Tilghman. Your suggestion did it. I ran into similar compile problem later: - /usr/src/zaptel-1.4.5.1/xpp/xbus-sysfs.c:135: error: unknown field âhotplugâ specified in initializer make[4]: *** [/usr/src/zaptel-1.4.5.1/xpp/xbus-sysfs.o] Error 1 - and I went in and

Re: [asterisk-users] DTMF signalling, SIP, and Background()

Am Mittwoch, den 26.09.2007, 11:08 +0200 schrieb Bastian Friedrich: > Hi, > > I am currently setting up a voice mail/IVR machine with asterisk (1.4.10 > at the moment). During testing and evaluation, all was fine; in the - > slightly different - production environment, the IVR contexts do not >

[asterisk-users] DTMF signalling, SIP, and Background()

Hi, I am currently setting up a voice mail/IVR machine with asterisk (1.4.10 at the moment). During testing and evaluation, all was fine; in the - slightly different - production environment, the IVR contexts do not react sensibly. The environment is: POTS <-- (ISDN) --> PBX <-- (SIP) --> Aste

Re: [asterisk-users] Do I need to run #modprobe zaptel for Digium

On Wed, Sep 26, 2007 at 12:13:33AM -0700, bilal ghayyad wrote: > Hi Cohen; > > And do I need to run #modprobe wcfxs / #modprobe wcfxs > or I have to run #modprobe wctdm? What is the > difference? Just use wctdm (modprobe wcfxs will likely have the same effect. wcfxs was the driver for the same

Re: [asterisk-users] Multiple Home system with SIP

I answered because I was hoping for a repost without the licence, perhaps through gmail. Would you have been happier not knowing that you were missing out on something? /Benny ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astric

Re: [asterisk-users] Do I need to run #modprobe zaptel for Digium

Hi Cohen; And do I need to run #modprobe wcfxs / #modprobe wcfxs or I have to run #modprobe wctdm? What is the difference? Regards Bilal > Hi List; > > If I am configuring Diguim Analoge card, then I need > to run #modprobe wctdm, but the question why I need to > run #modprobe zaptel also? N

[asterisk-users] Music On Hold

Hi All, I need to have the same file played from MoH every time someone gets to MoH from a Dial. I want to play marketing messages from it and I want it to start from file 1 every time. Anyone know if/how this can be done? Cheers, Joel. ___