Re: [asterisk-users] [OT] wi-fi ip phone scenario

Alban ha scritto: I've made some tests with Hitachi WIP3000 and 5000, works really good with roaming (without authentification). Some parts of the AP in the mesh are wired (no WDS), some others are not (using WDS), but all use the same SSID and channel. In all cases roaming was fast, quite not

[asterisk-users] Re: Linksys PAP2: calling tone stops after 5

Message: 7 Date: Sun, 29 Oct 2006 22:00:22 +0100 From: "Jose Limeres" <[EMAIL PROTECTED]> Subject: [asterisk-users] Linksys PAP2: calling tone stops after 5 tones To: asterisk-users@lists.digium.com Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; for

[asterisk-users] Call from internal num. to VoIP gate

Greetings to All! Help to solve a problem: There is an asterisk and two VoIP a sluice (NSGate 800 2FXS 2FXO, NSGate 800 2FXS). In sip.conf they are registered so: [3301] type=friend host=172.222.135.11 username=3301 secret= defaultip=172.222.135.11 dtmfmode=rfc2833 context=it callerid="V

Re: [asterisk-users] [OT] wi-fi ip phone scenario

I've made some tests with Hitachi WIP3000 and 5000, works really good with roaming (without authentification). Some parts of the AP in the mesh are wired (no WDS), some others are not (using WDS), but all use the same SSID and channel. In all cases roaming was fast, quite not possible to "hear"

Re: [asterisk-users] Incorrect Ring tone. Getting a US tone when it should be AU tone

What phones are you using? It could be a phone level issue. (my aastra has a setting for AU sounds..) PaulH On Mon, 2006-10-30 at 16:13 +1100, Klaverstyn, David C wrote: > For some reason Asterisk is producing a US ring tone when it should be > an Australian ring tone. I am using ztdummy and do

Re: [asterisk-users] No ring tone when using IAX

On 22:34, Sun 29 Oct 06, Al Bochter wrote: > When I call from a softphone using IAX2 there is no ring tone > This is the same if I call in to the IVR and press # and dial the > stations ext number no ring tone > And I get the same if I call in using a DID on an IAX2 trunk > > BUT > > if I use an

Re: [asterisk-users] Pager Voicemail Message

Hi Lacy,Thanks a lot.I may have deleted these lines from the voicemail.conf file in the past. I followed your suggestion and found these lines in the sample configuration file.Again, thank you very much.David- Original Message From: Lacy Moore - Aspendora <[EMAIL PROTECTED]>To: Asterisk Use

RE: [asterisk-users] Incorrect Ring tone. Getting a US tone when it should be AU tone

By default, the file contained the following. I'm guessing this is what you mean. [au] ringcadence = 400,200,400,2000 dial = 413+438 busy = 425/375,0/375 ring = 413+438/400,0/200,413+438/400,0/2000 congestion = 425/375,0/375,420/375,0/375 callwaiting = 425/200,0/200,425/200,0/4400 dialrecall = 41

Re: [asterisk-users] Pager Voicemail Message

On 10/29/06, David <[EMAIL PROTECTED]> wrote: I looked. There's nothing there.I even did a search under /etc/asterisk for files containing "Asterisk PBX" and "New VM" (both part of the Pager message) and it didn't help. I suspect that it may be in the code.   I would suggest looking again.  If

Re: [asterisk-users] CID and CDR conflict?

Mike Diehl wrote: Hi all, I've been beating my head against this for some time now. For incoming calls, I'd like to send my users a "localized" caller id number. By "localized," I mean one with out the 1+areacode for local calls and only 10 digits (minus the leading 1) for long distance call

Re: [asterisk-users] dialing external number within meetme

Use the manager API and send a call from the exten of the meetme room to the number you want to dial using the Local/ channel. MATT--- On 10/28/06, Bartosz Wegrzyn - maillists <[EMAIL PROTECTED]> wrote: hello, is it possible to dial out external number within running conference, for example di

Re: [asterisk-users] asterisk-1.2.13 fails to 'Make' in Fedore Core 6'

On Sun, Oct 29, 2006 at 07:53:21PM -0600, Ward, Bill wrote: > If I remember right, FC6 doesn't necessarily install the kernel source. Actually, the kernel headers are never required to build asterisk . /usr/include/linux is not from the kernel source. > I had this error before when I installed

Re: [asterisk-users] Incorrect Ring tone. Getting a US tone when it should be AU tone

On Mon, Oct 30, 2006 at 04:13:12PM +1100, Klaverstyn, David C wrote: > For some reason Asterisk is producing a US ring tone when it should be > an Australian ring tone. I am using ztdummy and do not have any cards > installed. My configuration is as follows. I am using Trixbox 1.2.2. > Can someo

[asterisk-users] CID and CDR conflict?

Hi all, I've been beating my head against this for some time now. For incoming calls, I'd like to send my users a "localized" caller id number. By "localized," I mean one with out the 1+areacode for local calls and only 10 digits (minus the leading 1) for long distance calls. For example: I

[asterisk-users] Incorrect Ring tone. Getting a US tone when it should be AU tone

For some reason Asterisk is producing a US ring tone when it should be an Australian ring tone.  I am using ztdummy and do not have any cards installed.  My configuration is as follows.  I am using Trixbox 1.2.2.   Can someone please guide me into the right direction?    zaptel.conf loa

[asterisk-users] Maximum talktime in a queue?

On 10/26/06, Lenz <[EMAIL PROTECTED]> wrote: When you log in a callback agent, you enter first the agent code, and then the extension he's sitting at. The context is usually specified in the dialplan command, but the result is that asterisk knows that agent 103 is sitting at [EMAIL PROTECTED] [EM

[asterisk-users] No ring tone when using IAX

When I call from a softphone using IAX2 there is no ring tone This is the same if I call in to the IVR and press # and dial the stations ext number no ring tone And I get the same if I call in using a DID on an IAX2 trunk BUT if I use anything that is SIP I get the ring tone Softphone DISA Tru

Re: [asterisk-users] Pager Voicemail Message

I looked. There's nothing there.I even did a search under /etc/asterisk for files containing "Asterisk PBX" and "New VM" (both part of the Pager message) and it didn't help. I suspect that it may be in the code.- Original Message From: Dovid B <[EMAIL PROTECTED]>To: Asterisk Users Mailing L

Re: [asterisk-users] Trixbox installation - ZAP channels becoming upresponsive

One of the reasons, which I've experienced, is that the computer's processor gets over loaded. If its not a genuine server, then this will happen. A standard computer made for general home or office use will give trouble on zap channels when used with any hardware which does not have its own microp

Re: [asterisk-users] meet me

Does it say invalid conference number, or invalid password?   You need to have either digium hardware installed, or ztdummy installed. Otherwise conferencing will not work.   And also it is better to install trixbox instead of AAH. ___ --Bandwidth and Col

RE: [asterisk-users] asterisk-1.2.13 fails to 'Make' in Fedore Core 6'

If I remember right, FC6 doesn't necessarily install the kernel source. I had this error before when I installed on my Dell server with FC6. I had to run these commands and it got past that part: yum install kernel yum install kernel source Then reboot. -Original Message- From: [EMAI

[asterisk-users] Multiple dial macros at the same time

I am setting up an after-hours on-call system. Someone calls in and requests service, and while they listen to music on hold, we dial out to several people's cell phones and home phones. We don't know if they will be answered by the employee, or by voicemail or a spouse/relative/child/pet. So w

Re: [asterisk-users] asterisk-1.2.13 fails to 'Make' in Fedore Core 6'

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Keithy escribió: > > I fresh installed Fedora Core 6. > > I downloaded and untar the 'asterisk-1.2.13' into /usr/src/asterisk-1.2.13. > When I run ' make' > > I get: > > ... > ... > chan_phone.c:41:29: error: linux/compiler.h: No such file or directo

RE: [asterisk-users] blind transfers with IP Polycom 501

Thanks so mAre there any drawbacks to this? This is the digitmap I ended up using and it seems to work: [2-9]11|0T|011xxx.T|[0-1][2-9]x|[2-9]x|[1-9]xxxT -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian J. M. Sent: Sunday, October 2

[asterisk-users] asterisk-1.2.13 fails to 'Make' in Fedore Core 6'

Hi, I fresh installed Fedora Core 6. I downloaded and untar the 'asterisk-1.2.13' into /usr/src/asterisk-1.2.13. When I run ' make' I get: ... ... chan_phone.c:41:29: error: linux/compiler.h: No such file or directory make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/usr/src/a

Re: [asterisk-users] Re: [OT] wi-fi ip phone scenario

WIP 330 and it sux :( - Original Message - From: "Alberto Pastore" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Sunday, October 29, 2006 11:35 AM Subject: [asterisk-users] Re: [OT] wi-fi ip phone scenario Martin Joseph wrote: I think it's c

Re: [asterisk-users] Pager Voicemail Message

Yes. It should be in that same file. Poke around. - Original Message - From: David To: asterisk-users@lists.digium.com Sent: Sunday, October 29, 2006 12:43 PM Subject: [asterisk-users] Pager Voicemail Message Hello,In voicemail.conf, it's possible to ed

Re: [asterisk-users] tx_fax not getting entire fax

Hi Jerry,I have never managed to get rx/tx fax working reliably. IAXModem (which in effect grafts steves spandsp library to hylafax) however does work well.-bradley On 10/29/06, Jerry Geis <[EMAIL PROTECTED]> wrote: Steve,I am trying to get tx_fax to work. I am using a TDM2401E card.I have a 3 page

[asterisk-users] Polycom IP500 Problems

Hi, Im having problems getting a polycom IP500 phone to register. Ive edited the xml files and appropriate .cfg files and it shows up registered with no ip address and if I go into the phones status menu and pick lines it says its not registered. If I tell it to upload the log file it doesn't say

Re: [asterisk-users] blind transfers with IP Polycom 501

Yes, digitmap... If you just want to allow any digit pattern, use this digitmap: xx.T x -> Any valid digit . -> 0 or more ocurences of previous charracter T -> Default timeout (3 seconds) Any digit followed by a 3 second timeout will match. You can include pattern to match * and #. xx.T|*x.T|#

RE: [asterisk-users] blind transfers with IP Polycom 501

It was the digit map. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeronimo Romero Sent: Sunday, October 29, 2006 6:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] blind transfers with IP Polycom

Re: [asterisk-users] Advice on GUI

Without providing a link to the list, or citing your front-runners, you can't really expect people to reply, can you?On 10/27/06, Frédéric Blaise < [EMAIL PROTECTED]> wrote:Hello allI would like to know your opinions on free GUI used to manage Asterisk. Which is better?My setup is quite small, abou

RE: [asterisk-users] blind transfers with IP Polycom 501

Do you mean the digitmap?? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Sunday, October 29, 2006 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] blind transfers with IP Polycom 501 For the sof

Re: [asterisk-users] blind transfers with IP Polycom 501

For the soft buttons to work the way you want it, make sure you got the Polycom dialplan setup right. On 10/29/06, Jeronimo Romero <[EMAIL PROTECTED]> wrote: I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only problem I'm experiencing is the following: I can't seem to g

[asterisk-users] blind transfers with IP Polycom 501

I’m running Asterisk 1.2.8 with Polycom ip501’s xten softphones  The only problem I’m experiencing is the following: I can’t seem to get blind transfers to work with my Polycom 501 phones  Either through the feature code or the soft keys.       Feature code blind transfers: I set up a

Re: [asterisk-users] SIP v IAX2

I had a slide on this at my astricon presentation last week (about developing our IAX/java softphone) The slide basically said that I did IAX instead of SIP because the IAX draft RFC is ~100 pages whereas there are > 100 RFCs on SIP/STUN/RTP - That's a lot more reading ! (Plus of course

[asterisk-users] AEL2 and the variables

Hi, I am using Asterisk 1.2.12.1 + the AEL2 patch. If I use a variable instead of the extension itself, an incoming call cannot be connected. ${ID-FST1} => Dial(SIP/gs|15|r); <== NON ok sip debug shows : Looking for 6674262730 in interne (domain 192.168.1.14) SIP/2.0 404 Not Found I

[asterisk-users] Asterisk Voicemail with ODBC Realtime Access

Hi I was trying to have realtime voicemail working with ODBC Driver.Everything works fine with MySQL Realtime access, BUT as I want to implement ODBC Storage as well, it seems that everything should go through ODBC ( what I read on voip-info wiki page ) But I do not manage to make it work with ODBC

Re: [asterisk-users] No zap* commands?

On Sun, Oct 29, 2006 at 04:07:21PM -0500, Jim Lynch wrote: > Conrad Wood wrote: > > > >On 29 Oct 2006, at 20:24, Jim Lynch wrote: > > > >>I've compiled and installed the zap modules but asterisk still > >>doesn't show any zap commands when I do a help. Any suggestions as > >>to why? > >> > >> >

Re: [asterisk-users] Out bound calls 'you must first dial a 1'

George Patterson wrote: Hello, I have asterisk 1.2.9 running on a Debian sarge server, my outbound dial plan looks something like this: [outbound-longdistance] exten => _91NXXNXX,1,Dial(${OUTBOUND1}/${EXTEN:1}) This has been covered quite a bit in the archives. Some providers want a s

Re: [asterisk-users] Something is trashing /var/run

Rodrigo Gonzalez wrote: mkdir /var/run/asterisk in /etc/asterisk/asterisk.conf change where you see /var/run with /var/run/asterisk Jim Lynch wrote: For some reason, asterisk is changing ownership of all the files in /var/run to itself. /var/run/* now belongs to asterisk.asterisk after a r

Re: [asterisk-users] No zap* commands?

Conrad Wood wrote: On 29 Oct 2006, at 20:24, Jim Lynch wrote: I've compiled and installed the zap modules but asterisk still doesn't show any zap commands when I do a help. Any suggestions as to why? zap modules not loaded? try: load chan_zap.so on the console and/or put that into modul

Re: [asterisk-users] No zap* commands?

Jim Lynch wrote: > I've compiled and installed the zap modules but asterisk still doesn't > show any zap commands when I do a help. Any suggestions as to why? > only a hint have you restarted asterisk? or either, have you checked that all of zaptel &co has installed successfully? hope you find so

[asterisk-users] Linksys PAP2: calling tone stops after 5 tones

Hi all, I have a problem with the dialing tone in PAP2: When making a call, I can hear the calling tone 5 times and then it stops. The called party still hears the call but not the calling party. I've playing around with different parameters on the PAP2 web config with no success until now. Anyo

Re: [asterisk-users] Something is trashing /var/run

mkdir /var/run/asterisk in /etc/asterisk/asterisk.conf change where you see /var/run with /var/run/asterisk Jim Lynch wrote: For some reason, asterisk is changing ownership of all the files in /var/run to itself. /var/run/* now belongs to asterisk.asterisk after a reboot. I installed zapt

[asterisk-users] Out bound calls 'you must first dial a 1'

Hello, I have asterisk 1.2.9 running on a Debian sarge server, my outbound dial plan looks something like this: [outbound-longdistance] exten => _91NXXNXX,1,Dial(${OUTBOUND1}/${EXTEN:1}) About every other outbound call we make, we get the 'you must first dial a 1' message from our phone

[asterisk-users] Something is trashing /var/run

For some reason, asterisk is changing ownership of all the files in /var/run to itself. /var/run/* now belongs to asterisk.asterisk after a reboot. I installed zaptel, wanpipe and asterisk on a fresh install of CentOS. I did the same thing on Friday and had the same problem, so I scrubbed t

Re: [asterisk-users] No zap* commands?

On 29 Oct 2006, at 20:24, Jim Lynch wrote: I've compiled and installed the zap modules but asterisk still doesn't show any zap commands when I do a help. Any suggestions as to why? zap modules not loaded? try: load chan_zap.so on the console and/or put that into modules.conf

[asterisk-users] No zap* commands?

I've compiled and installed the zap modules but asterisk still doesn't show any zap commands when I do a help. Any suggestions as to why? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update

Re: [asterisk-users] Diva server 4bri and Portuguese BRI

Finally this works!!! :) Tanks to Alberto and Marco by your help! The problems are: - the cable was connected to the wong card port... :( - the card config needs to be: ETSI; TE; Point-to-Point (I thought that was point-to-multipoint). Best regards, PS. 2006/10/29, Pedro Silva <[EMAIL PROTECTED]

[asterisk-users] hardware requirements..

Hello, I have Asterisk running on Debian Sarge. I use AGI to billing and now I´m planing to separate mysql from Asterisk Box, resulting: (Asterisk + AGI) - (Iptables + showrewall + Apache + PHP + MySQL) What hardware configuration do you recommend for the second box, in order to maintain

[asterisk-users] H.263 Video Messages

Hello,I'm trying to set the Asterisk to leave a video message to the mailbox, but there is some compatibility problem, although h263 is identified as the matching codec, as you can see in the debug messages below:Capabilities: us - 0x80100 (g729|h263), peer - audio=0x43f (g723|gsm|ulaw|alaw|g726|ad

Re: [asterisk-users] Diva server 4bri and Portuguese BRI

Hello again Alberto! Anyway, to get more info, try to open a second shell and run /usr/lib/eicon/divas/xlog then on the first shell redo the telsampl test, then post the output of xlog off the list to my address (alberto at msoft-italia.com) This is the xlog output: 4:1736:074 - CREATEID ok: c

Re: [asterisk-users] Diva server 4bri and Portuguese BRI

Olá Marco! :) 2006/10/29, Marco Mouta <[EMAIL PROTECTED]>: pls post your misdn.conf as well as extensions.conf The asterisk version that i used with trixbox dont't have misdn.conf... I used capi.conf. For now, i dont care about asterisk, because with the divas utility telsampl i know that the

Re: [asterisk-users] VoIP GSM Gateways

Hi, On Sun, 2006-10-29 at 13:46 +0200, Tzafrir Cohen wrote: > Is vISDN (extra kernel modules, extra non-standard Asterisk channel) > required? The page on vGSM there suggests it is. no, vgsm uses only a part of visdn (timer system and streamport), so you need only chan_vgsm, visdn_streamport (fo

Re: [asterisk-users] Diva server 4bri and Portuguese BRI

Marco Mouta ha scritto: pls post your misdn.conf as well as extensions.conf May be i can help. Sou Português:) On 10/29/06, *Pedro Silva* < [EMAIL PROTECTED] > wrote: Thanks Alberto! I tested with telsampl like you said (with various configurations for

Re: [asterisk-users] Asterisk Manager

Yes, I did check it.  The asterisk.st1 is running at /var/run folder, I tried to change to /var/run/asterisk/asterisk.ct1, then I will have the error "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?"   I thing it have something with the permission.  How can

Re: [asterisk-users] Diva server 4bri and Portuguese BRI

pls post your misdn.conf as well as extensions.confMay be i can help.Sou Português:)On 10/29/06, Pedro Silva < [EMAIL PROTECTED]> wrote:Thanks Alberto!I tested with telsampl like you said (with various configurations for de diva) and this not works...:(The trace is:Enter destination address: 273xxx

Re: [asterisk-users] Diva server 4bri and Portuguese BRI

Thanks Alberto! I tested with telsampl like you said (with various configurations for de diva) and this not works...:( The trace is: Enter destination address: 273xx <--Conn_Req(273xx) Connect_Con--> [29]:Disc_Ind--> <--Disc_Res **Call cleared*** Any idea for the possible

Re: [asterisk-users] VoIP GSM Gateways

2006/10/29, Forum <[EMAIL PROTECTED]>: Peter,How much does the 4 port cost? How many simultaneous calls can you make? Doyou need a mobile account from a mobile provider such as T-mobile?I've been told it costs 1600 Euros for 4 ports and 1400 for 2 ports. Regards

Re: [asterisk-users] VoIP GSM Gateways

On Sun, Oct 29, 2006 at 01:27:25PM +0100, Matteo Brancaleoni wrote: > Hi, > > On Sun, 2006-10-29 at 09:16 +0100, Michiel van Baak wrote: > > On 18:15, Sat 28 Oct 06, Forum wrote: > > > I'm looking at setting up a VoIP GSM gateway to connect to my asterisk > > > box. > > > What experience have peo

Re: [asterisk-users] VoIP GSM Gateways

Hi, On Sun, 2006-10-29 at 09:16 +0100, Michiel van Baak wrote: > On 18:15, Sat 28 Oct 06, Forum wrote: > > I'm looking at setting up a VoIP GSM gateway to connect to my asterisk box. > > What experience have people on this list have with GSM gateway hardware. I > > have been looking at the 2N voic

Re: [asterisk-users] porting numbers in UK telewest/bt/adept

On 26 Oct 2006, at 11:59, Conrad Wood wrote: A client used to use BT isdn30 and ported the numbers to telewest several years ago. Now, the client moved to adept telecom. I *think* adept resells BT products. We got new numbers from adept (bt?) and the old pbx on the telewest lines forwards the c

[asterisk-users] Pager Voicemail Message

Hello,In voicemail.conf, it's possible to edit the voicemail message, but when I define a pager email address, I get the message from "Asterisk PBX", and the content is fixed by the system.Is there a way to manipulate this message, as well?Thanks,David___

Re: [asterisk-users] app_meetme not loading

On Sun, Oct 29, 2006 at 04:47:48PM +0800, Will Roy wrote: > I originally built my Asterisk server without installing the Zaptel package > as it was going to be a purely SIP based system. However when I went to > setup conferencing using meetme I found out that app_meetme is dependant on > the ztdum

Re: [asterisk-users] Re: Voicemail and OSX 10.4 Intel

Well, yes I'm sorry I'm using 1.2.13, all compile is ok, also I've installed mpg123. all modules loaded fine, and all codecs too. One thing keep my attention and is that when I change format for recording, it ever uses wav|wav49, I tried to change on voicemail.conf format to only gsm, but is not r

Re: [asterisk-users] Asterisk behind NAT and without portforwarding forrtp

Am Sunday 29 October 2006 01:31 schrieb Dovid B: > Half asleep. Sorry for my last post. I believe you still need port > forwarding for IAX. Time to keep to my bed time. If works as long as you have notransfer=no at both ends. Iam concerned that with SIP Asterisk is bridging up and I do not receiv

Re: [asterisk-users] translate.c:88 powerof: Powerof 0: No power??

2006/10/28, Moises Silva <[EMAIL PROTECTED]>: This is a problem of the codec you are attempting to use. Wich codecis?, it seems to Asterisk cannot identify the codec you are using.RegardsOn 10/28/06, Giedrius Augys < [EMAIL PROTECTED]> wrote:> Hi,>  I have just installed fresh FreeBSD 6.1 and ast

[asterisk-users] app_meetme not loading

I originally built my Asterisk server without installing the Zaptel package as it was going to be a purely SIP based system. However when I went to setup conferencing using meetme I found out that app_meetme is dependant on the ztdummy for timing. I have now installed the zaptel package and I belie

Re: [Asterisk-Users] Would you support a Bristuff mailing list ?

Hi,Up to now, I can think of 6 people claming interest in such list.As bristuff somehow modifies Asterisk, people using bristuff and asking support in Asterisk User list will be forwarded to this list.So this number should be growing. Maybe, a core of 20 people would be enough to turn this idea int

[asterisk-users] Re: [OT] wi-fi ip phone scenario

Martin Joseph wrote: I think it's cleary true that wiring WIFI infrastructure is easier and more reliable then WDS. On the other hand, I have been running my little network with WDS for over three weeks now, and it has been completely reliable. The tricks where to configure things properly

Re: [asterisk-users] Compiling Zaptel 1.2.10 on Ubuntu 6.10

Dear Storm, I have two guesses One could be something in the ubuntu make which makes it unable to understand some regx in the scripts used or I am not quite sure but check the kernel version you are having (i do that by uname -a ) I believe you will find something there, if it is not the same as

Re: [asterisk-users] Diva server 4bri and Portuguese BRI

Pedro Silva ha scritto: Hello, I need to connect one diva server 4bri to a portuguese BRI interface. The operator (PT) said that this bri is in point-to-multipoint mode (S0). Previously one PBX has connected to that interface. The asterisk and diva drivers are working ok but i cannot communicate

Re: [asterisk-users] How to make different ext using different trunks?

Put them in diffrent context's and then have a seperate context allowing everyone to talk to each other (locally) and have an include in every context. - Original Message - From: Zeeshan Zakaria To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sund

Re: [asterisk-users] VoIP GSM Gateways

On 18:15, Sat 28 Oct 06, Forum wrote: > I'm looking at setting up a VoIP GSM gateway to connect to my asterisk box. > What experience have people on this list have with GSM gateway hardware. I > have been looking at the 2N voiceblue products. Hi, We are using a voiceblue in our office. It's a voi