Re: [Asterisk-Users] BUSY-tone on incoming calls?

On Tue, 25 Jan 2005, Peter Svensson wrote: On Tue, 25 Jan 2005, Tobias Jönsson wrote: No, PRI_CAUSE works great at least in 1.0.2, probably in the earlier 1.0 releases too. Busy() may play a busy tone to the caller instead of signalling busy so using PRI_CAUSE is much better in PRI or BRI enviro

Re: [Asterisk-Users] cant do it in CLI anymore?

Hi, Do you have load chan_oss/chan_alsa ? if not then you can't use 'Dial' app. Hope this help. :) Mick Hastings on 2005/1/26 03:31 wrote: >Hi Floks, > > > >*CLI> Dial >No such command 'Dial' (type 'help' for help) >*CLI> > > >the same thing for Answer, Hangup, etc > >what have I missed? > >chee

RE: [Asterisk-Users] Asterisk calls back after phone call

I upgraded to firmware version x.22 and it went away. I think it is a bug in the phone. On Tue, 2005-01-25 at 10:55 +0200, Doug Reid - Stormcorp wrote: > I get the same thing. Its as if the grandstream does'nt > send a hangup signal. > > Someone out there must have fixed this??? > > Doug > >

[Asterisk-Users] cant do it in CLI anymore?

Hi Floks, This is probably really dumb but here goes: I used to be able to place calls to my SIP phones from the CLI using the 'Dial' command for testing. I have installed asterisk on a new machine and copied over the .confs and started it up. It all works fine. But when I try to initiate a ca

Re: [Asterisk-Users] size and quality of audio clips effecttheplayback??

On Tue, 2005-01-25 at 23:14 -0800, Gabriel Afana wrote: > Hi, > This is what I am running: Red Hat Enterprise Linux ES release 3 > (Taroon Update 4) > > Is the Taroon the kernel version? Do you think this could be a kernal issue > (did you hear it for yourself at the site)? No Taroon I woul

Re: [Asterisk-Users] Problems splicing Asterisk with a TE405P between Arcor E1 PRI and Ericsson Business Phone 250

On Wed, 26 Jan 2005, Klaus-Peter Junghanns wrote: > Am Dienstag, den 25.01.2005, 22:39 +0100 schrieb Frank Sautter: > > the setup before: > > Arcor TelCo PRI(E1) <> Ericsson BP250 PRI(E1) > > > > the setup desired with asterisk spliced in: > > Arcor TelCo PRI(E1) <> P1

Re: [Asterisk-Users] SOLVED: Wireless Grandstream via NATing wifi laptop

I fixed the problem I had with my laptop crashing after I hung up from a call. The problem appeared to be caused by running ndiswrapper (the driver wrapper for the wireless card in my laptop) with a 4K stack in a 2.6.10 kernel. I rebuilt the kernel with the 4K stack disabled and the problem disa

RE: [Asterisk-Users] Florz patch for zaphfc

Stuart, Can you plase specify in which mode are you using your hfc cards ? You said ptp, but are they working as NT or TE ? Thanks, Ivan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stuart Hirst Sent: Wednesday, January 26, 2005 12:26 AM To: asterisk-use

Re: [Asterisk-Users] Tall free number via FWD over IXA2

Joseph wrote: Thanks Kris, I found the solution: Here is how it suppose to look like: You can minimise all that with a simple macro and a little pattern matching, and it makes dial plans so much easier to track down problems with etc... I couldn't find anything on it, but I'm not sure if you can

Re: [Asterisk-Users] size and quality of audio clips effecttheplayback??

Hi, This is what I am running: Red Hat Enterprise Linux ES release 3 (Taroon Update 4) Is the Taroon the kernel version? Do you think this could be a kernal issue (did you hear it for yourself at the site)? Gabe - Original Message - From: "Steven Critchfield" <[EMAIL PROTECTED]>

Re: [Asterisk-Users] Asterisk with Grandstream ringback

I updated to firmware version x.22 and this and a few other problems were fixed. I was running x.18 and it allowed me to do a successful upgrade via http. On Tue, 2005-01-25 at 08:10 -0700, Kim Lux wrote: > Are you saying that you are running firmware X.22 and it is not doing > the callback wh

Re: [Asterisk-Users] Damn DTMF Beeps on my calls

Well this happens a LOT when I call one particular person, not so much when I call others. Both sides of the call are running Sipura ATA's with * in the middle, no termination or Zap in between at all. It seems that when I call this person from my home address it occurs a LOT like 1 or 2 times

Re: [Asterisk-Users] Tall free number via FWD over IXA2

On Wed, 2005-01-26 at 00:17 -0500, Kris Stark wrote: > Joseph wrote: > > I've setup my IAX2 over FWD and it is working I can receive a test call > > and I can call out. > > Though I cannot figure out how to dial 1-800 numbers over FWD > > When I dial 1-800 it hangs up on me. > > > > Here is a typi

Re: [Asterisk-Users] Any experience with Sangoma cards? and R2MF

On Tue, 25 Jan 2005 23:35:47 -0600, Fernando Romo wrote: > Here in Mexico the R2MF is a "standard" for the dominat phone company > (Telmex), The Sangoma People tell me the products must work with the R2 > librarys wrote by Steve Underwood. Should work exactly the same as digium cards, i think it u

Re: [Asterisk-Users] Any experience with Sangoma cards? and R2MF

Any have a Samgoma E1 working with R2MF signalling? Here in Mexico the R2MF is a "standard" for the dominat phone company (Telmex), The Sangoma People tell me the products must work with the R2 librarys wrote by Steve Underwood. Any expierence out there? Thanks in advanced.. Fernando Romo

[Asterisk-Users] New RPMS for FC1

I have uploaded a new FC1 kernel, ALSA and asterisk-v1.0.5 to the same old place. ftp://ftp.linuxsys.com/pub/releases/FC1/alsa ftp://ftp.linuxsys.com/pub/releases/FC1/kernel ftp://ftp.linuxsys.com/pub/releases/FC1/kernel-smp ftp://ftp.linuxsys.com/pub/releases/FC1/

Re: [Asterisk-Users] Tall free number via FWD over IXA2

Joseph wrote: I've setup my IAX2 over FWD and it is working I can receive a test call and I can call out. Though I cannot figure out how to dial 1-800 numbers over FWD When I dial 1-800 it hangs up on me. Here is a typical session: Called x:[EMAIL PROTECTED]/18007425877l -- Call accepted by

RE: [Asterisk-Users] Asterisk@Home initial setup

Hi Jason, Doesn't sounds like it. I run a P3 -750 and takes about 15 minutes to install. I don't have advice on what else you can do. Just letting you know a P@ 400 might be a little underpowered but should work. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL

[Asterisk-Users] Caller ID w/Name Providers???

Do any IAX or SIP providers besides Broadvoice offer Caller ID with Name on INCOMING calls yet? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update option

Re: [Asterisk-Users] BroadVoice Help

> .. but I don't get incoming calls anymore What happens on incoming calls then? Is the caller sent to BV's mailbox? Does SIP debug show any incoming INVITE's from Broadvoice? --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com ht

Re: [Asterisk-Users] Damn DTMF Beeps on my calls

Hi Chuji, this reply is a bit late, but I had to try a few things to confirm. I managed to capture one of those sporadic beeps in a voicemail (which in my case are not DTMF tones, but just VERY loud squeaks). I had * record the voicemail as uLAW, G726-32 and GSM (the original caller used G726-32

[Asterisk-Users] Asterisk@Home initial setup

Hello, I'm attempting to learn about Asterisk, and I figured the easiest way is to use [EMAIL PROTECTED] to make it an easy install. After CentOS loads and the cd is auto ejected, my computer then reboots. However, on reboot all the farther I get is when it's trying to load GRUB it locks up the c

[Asterisk-Users] softphone headsets

Anybody have a suggestion for a nice inexpensive headset for mobile users on a laptop with a softphone? What are people using for softphones on M$ platforms? I have been using the x-lite client. Is there something better out there? -- http://www.umich2.com ___

[Asterisk-Users] Another BroadVoice Problem

I finally got my incoming and outgoing working but outgoing I cannot hear the called person, but the called person can hear me.   On incoming everything works perfect.     ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com h

[Asterisk-Users] SIP clients and double NAT

  Hi Asterisk users   I have a problem with a configuration as shown in the diagram…   +-+ | PBX/STUN server | +-+     |     |     | +-+ | FIREWALL & NAT  | +-+     |     |     | +--

Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone

> >thanks for info. Which iax softphones are using newer iaxclient ? What is > >the best iax softphone from this point of view ? > > > > > I don't know for sure, but I think iaxcomm and DIAX are most up-to-date. I'm almost finished building my IAX softphone that is based on a recent version of iax

RE: [Asterisk-Users] BroadVoice Help

Well I got to place outbound calls without the patch (iam running 1.0.5) Just had to change host = proxy.lax.sip.broadvoice.com to host=sip.broadvoice.com .. but I don't get incoming calls anymore And all those previous refunds not being issued threads are scaring me. -Original

[Asterisk-Users] TDM400 - channel out to lunch?

Today I had channel 1 on my TDM400 go to sleep, verified by pluggin in known good telco lines in various combinations on channel 1 through 4 - problem is channel 1, not anything external. So after seeing lots of stuff on the list re: TDM400's I power cycled, removed board and let linux say noth

[Asterisk-Users] Bellster enum

Ed Guy wrote: * ENUM directory. (server side is done -- hopefully someone will donate the client side.) see http://www.bellster.net/web/NewFeatures Umm there kind of is a working enum client side, it even does ownership checking on numbers before accepting them... :) http://www.e164.org What is

Re: [Asterisk-Users] BroadVoice Help

On Tue, 25 Jan 2005 18:34:42 -0800, John Sawa <[EMAIL PROTECTED]> wrote: > I had this same problem when I was setting up my * box, and it was due > to the BroadVoice patch not being applied. I had to get the Asterisk > source 1.0.3 then apply the patch and recompile. The reason you are > seeing the

[Asterisk-Users] Tall free number via FWD over IXA2

I've setup my IAX2 over FWD and it is working I can receive a test call and I can call out. Though I cannot figure out how to dial 1-800 numbers over FWD When I dial 1-800 it hangs up on me. Here is a typical session: Called x:[EMAIL PROTECTED]/18007425877 -- Call accepted by 65.39.205.12

RE: [Asterisk-Users] Re: Autio cut off at beginning of call

I'm running Asterisk 1.0.5 stable. Before that I have run 1.0.3, 1.0.1, 1.0 and many CVS versions all with the same symptoms.   Thank you, Reid Forrest, CISSP Max-IS, Inc. [EMAIL PROTECTED] ofc: 407.786.9600 x1200   cell: 321.439.8903   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTE

Re: [Asterisk-Users] Polycom 1.4.1 firmware for IP500/IP600

I'm looking for a copy too. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: "Robert Augustyn" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Tuesday, January 25, 2005 5:00 PM Subject: RE: [Asterisk-Users] Polycom 1.4.1 firm

[Asterisk-Users] Perfect billing solution for *?

Hi, I wonder what would you consider a perfect billing solution for *? What would it have to have and what would be nice robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users

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

Does anyone know if [EMAIL PROTECTED], which is running Asterisk CVS-v1-0-01/18/05-11:35:19, can be upgraded to 1.0.3 or higher? Jeff -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brian Dingman Sent: Tuesday, January 25, 2005 9:41 PM To: Asterisk Users Ma

RE: [Asterisk-Users] Polycom 1.4.1 firmware for IP500/IP600

If you have it, can I get a copy please, or possibly can you send it to the keeper of http://www.freedomphones.net/polycom/files/ I am looking for the latest boot image too. Thanks. robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Devenijn S

Re: [Asterisk-Users] Asterisk@home with Wildcard TDM400P card.

I think you need to modify the /etc/zaptel.conf file and the /etc/asterisk/zapata.conf file. - Original Message - From: "Chuck Keeter" <[EMAIL PROTECTED]> To: Sent: Tuesday, January 25, 2005 8:39 PM Subject: [Asterisk-Users] [EMAIL PROTECTED] with Wildcard TDM400P card. Hi all, I post

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

LiveVoip has a problem with Asterisk users on versions less than 1.0.3 If you are not using that version you need to upgrade now. We have a problem with two of our carriers at their gateway related to the Asterisk users. Our staff has developed a patch that is being tested at this time. Once the p

[Asterisk-Users] Asterisk@home with Wildcard TDM400P card.

Hi all, I posted this to the list yesterday, but not sure it made it. I'm running the latest [EMAIL PROTECTED] release, and am trying to install a Wildcard TDM400P with one FXS port and one FXO port. Linux see's the card during the start up ( I think) but when I modprobe for the card it's not b

Re: [Asterisk-Users] PrivacyManager not Working

So are you saying that * does not see the callerid but it should. Is this a possible bug in the callerid application. RIght now I am seeing that callerid isn't recognized 100% of the time (or possibly not transmitted) when I receive calls from VP Connect. If I do a NoOp(${CALLERIDNUM}) on incoming

Re: [Asterisk-Users] BroadVoice Help

I had this same problem when I was setting up my * box, and it was due to the BroadVoice patch not being applied. I had to get the Asterisk source 1.0.3 then apply the patch and recompile. The reason you are seeing the 404 is that asterisk is not registering to their service properly, and that

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

LiveVoip has developed a patch for a problem that our upstream gateway to Level 3 is having with Asterisk users. We hope to deploy this patch near the evening and are sorry for the delay. This problem stems from codec issues in Asterisk. You must use Asterisk ver. 1.0.3 as well. As soon as our engi

Re: [Asterisk-Users] Problems splicing Asterisk with a TE405P between Arcor E1 PRI and Ericsson Business Phone 250

Hi, please make a "pri debug span 2" of a call from the PBX to * and show us the contents. best regards Klaus -- Klaus-Peter Junghanns Am Dienstag, den 25.01.2005, 22:39 +0100 schrieb Frank Sautter: > hi, > > i'm having problems getting asterisk spliced between an E1 PRI (german > Telco Arco

Re: [Asterisk-Users] TE110P yellow errors

Hi All, I fixed the missed IRQ's with the following reference: http://lists.digium.com/pipermail/asterisk-users/2004-July/054712.html At 01:09 AM 26/01/2005, you wrote: Hi All, I have a TE110P in E1 mode, in a dell poweredge 250. The 30 channel E1 supplied is from a telco in Australia, with the f

[Asterisk-Users] Dial command announcement

The Dial command can be made to make an announcement to the called party before channel is bridged. Is it possible to make that announcement a Festival command in some way. -- Howard. LANNet Computing Associates; Your Linux people

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

Thanks Jeff! I think it's a little too late to find this info out. 3 to 4 days of no service. I have send many emails and still awaiting a response. Reminds me of my ILEC (QWEST) Do you have any info on what this patch does? -later On Tue, 2005-01-25 at 18:20, Jeff Glassman wrote: > They are

Re: [Asterisk-Users] fwd IAX2 error

> You defined the authentication to use the public key, > freeworlddialup.pub, and asterisk cannot find it. > > From the FWD web page: > > You will also need the freeworlddialup.pub key in your > /var/lib/asterisk/keys/ directory. (If you installed from cvs after June > 3, you probably have

Re: [Asterisk-Users] Updating Asterisk

Adam Robins wrote: There is no .version file anywhere in the /usr/src/asterisk tree -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, January 25, 2005 2:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: R

Re: [Asterisk-Users] Unable to Specify Channel 1 - no such device oraddress

When trying to start asterisk with a four-port FXS/FXO card I get the following error: "Unable to Specify Channel 1 - no such device or address" (Channel 1 should be the first FXO port). I looked in the asterisk-users mailing list archive and found one person who had this problem but did not fi

[Asterisk-Users] Problems splicing Asterisk with a TE405P between Arcor E1 PRI and Ericsson Business Phone 250

hi, i'm having problems getting asterisk spliced between an E1 PRI (german Telco Arcor) and an Ericsson Business Phone 250 digital PBX. The Asterisk Server has a TE405P with it's port 1 connected to the E1 PRI provided by our telecommunications provider Arcor and port 2 connected to the E1 PRI o

Re: [Asterisk-Users] Unable to Specify Channel 1 - no such device or address

Gene Naden wrote: When trying to start asterisk with a four-port FXS/FXO card I get the following error: "Unable to Specify Channel 1 - no such device or address" (Channel 1 should be the first FXO port). I looked in the asterisk-users mailing list archive and found one person who had this prob

Re: [Asterisk-Users] Your Acerbic Tyrant will be off line for about 10 days

Race Vanderdecken wrote: Greetings List, I know many of you are looking for advice from me but I am moving from the 28th until about the 4th of February. Race Vanderdecken Hmm, let me check my schedule... Sorry, no that doesn't work for me - you'll have to reschedule. Steve ___

RE: [Asterisk-Users] Updating Asterisk

On Tue, 2005-01-25 at 15:41 -0500, Adam Robins wrote: > There is no .version file anywhere in the /usr/src/asterisk tree How did you check. Are you aware that files starting with a period is considered a 'hidden' file. It doesn't show up in a ls entry unless you specify to show the files starting

[Asterisk-Users] DTMF digit dropping

I run an automated information retrieval system, using Asterisk. Fairly often the system misses a dialed digit. Our codes are all 4 digits, see lots of logs with: 4199 - OK 530 - Invalid code 330 - Invalid code 5330 - OK As callers experience skipped codes. We're usi

Re: [Asterisk-Users] SIP UDP ports on firewal to open

> I notice most things say to open ports 1-2 for UDP for SIP, > however from time to time this range isn't where Asterisk is opening the > ports: > > We're at xxx.xxx.xxx.xxx port 8542 > Answering with capability 0x2(GSM) > Answering with capability 0x4(ULAW) > Answering with capability 0x

Re: [Asterisk-Users] Problems splicing Asterisk with a TE405P between Arcor E1 PRI and Ericsson Business Phone 250

you have to experiment with the pridialplan= setting cheers Michael On Tue, 25 Jan 2005 22:39:14 +0100, Frank Sautter <[EMAIL PROTECTED]> wrote: > hi, > > i'm having problems getting asterisk spliced between an E1 PRI (german > Telco Arcor) and an Ericsson Business Phone 250 digital PBX. > The

Re: [Asterisk-Users] fwd IAX2 error

Joseph wrote: I'm trying to test IAX2 with FWD It registers fine but when I try to receive the call I get: chan_iax2.c:476 iax_error_output: Ignoring unknown information element 'Unknown IE' (38) of length 1 Jan 25 18:02:12 WARNING[114696]: chan_iax2.c:476 iax_error_output: Ignoring unknown inf

Re: [Asterisk-Users] fwd IAX2 error

On Tue, 2005-01-25 at 18:27 -0700, Joseph wrote: > I'm trying to test IAX2 with FWD > > It registers fine but when I try to receive the call I get: > > chan_iax2.c:476 iax_error_output: Ignoring unknown information element > 'Unknown IE' (38) of length 1 > Jan 25 18:02:12 WARNING[114696]: chan_

Re: [Asterisk-Users] AMP with SUSE 9.2

Same for me... No confirmation... Denis. Em Ter 25 Jan 2005 17:38, Keith Burns escreveu: > Ok, I signed up a few hours ago for the AMP mailing list, and no > confirmation. > > If anyone on this list has installed AMP with SUSE 9.2, if you wouldn't > mind emailing me with any gotchas at [EMAIL P

RE: [Asterisk-Users] TDM400 in aging Dell Optiplex

> -Original Message- > From: Jeff Pratt [mailto:[EMAIL PROTECTED] > Got one running in an Optiplex GX100. Works fine. I put one in an Optiplex GX170, for testing. In the topmost PCI slot, it couldn't generate interrupts. Worked in the next slot down, though. TDM400 + Dell seems to be

[Asterisk-Users] fwd IAX2 error

I'm trying to test IAX2 with FWD It registers fine but when I try to receive the call I get: chan_iax2.c:476 iax_error_output: Ignoring unknown information element 'Unknown IE' (38) of length 1 Jan 25 18:02:12 WARNING[114696]: chan_iax2.c:476 iax_error_output: Ignoring unknown information elem

Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone

- Original Message - From: <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, January 25, 2005 2:44 PM Subject: Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone > > (IAXPhone): > I suppose you're talking about Steve Sokol'

[Asterisk-Users] Re: Autio cut off at beginning of call

For what it is worth I am experiencing the exact same problem with the latest CVS.  I have tried numerous IAX providers and the problem follows so it isn’t the provider.   Are you running stable or CVS? I think that you hit it on the head, the fact that audio is being sent prior to an IAX ANS

Re: [Asterisk-Users] Cisco 7940/7960

Mark Johnson wrote: This may be OT, but I can't seem to find how to do this. I have 7940/7960's with Skinny on them. When you start pressing numbers on the dialpad, you start building a number to dial. When I install SIP, that functionality goes away. You have to hit the speaker button, or

[Asterisk-Users] Am i in control after i dial?

I want to be able to interrupt a conversation and let festival say something like "you have 10 seconds left for this conversation". Can i interrupt a call after its been dialed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.

Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone

Robert Rozman wrote: - Original Message - From: <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, January 25, 2005 2:44 PM Subject: Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone (IAXPhone):

RE: [Asterisk-Users] Updating Asterisk

1. I wiped out the /usr/src/asterisk directory structure 2. I followed the instructions below for re-downloading, installing and restarting Asterisk 3. The Asterisk module in /usr/sbin/asterisk reflects the new date/time Still shows version 1-0 12/21/2004. I can not find a .version file in

RE: [Asterisk-Users] BroadVoice Help

For me, yes, a lot better in many ways.   -Original Message- From: Chris Ford [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 25, 2005 8:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] BroadVoice Help   Does broadvoice offer b

Re: [Asterisk-Users] BroadVoice Help

Does broadvoice offer better service than voice pulse?   Chris FordCMF International Technologies LLC.[EMAIL PROTECTED]   - Original Message - From: Kevin Kiely To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, January 25, 2005 7:51 PM

[Asterisk-Users] calleridname from chan_sip (mysql_sipfriends)

Hi, I'm using mysql to define my sipfriends. When authenticating, the calleridname from the calling SIP user (phone) seems getting lost. With "sip debug" I can see in the SIP messages: From: "myName" ;tag=22668125 To: but I can't find "myName" in any channel variable. Both, ${CALLERID} and ${CALLE

RE: [Asterisk-Users] BroadVoice Help

Try this:   dtmfmode=inband register => [number]:[EMAIL PROTECTED]     [broadvoice] type=peer fromuser=[number] host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=from-broadvoice reinvite=no canreinvite=no pedantic=yes qualify=yes disallow=all allow=alaw     -

RE: [Asterisk-Users] AMP with SUSE 9.2

Yeah, I got the AMP part working but in the process messed up my * install WRT to the ZAP stuff and /dev (I *think* made some changes to the ZAP Makefile to support SUSE 9.2 and udev last time I installed, but didn't make those changes this time - I am a complete Linux noob). It was interesting tr

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

They are coming out with a patch for the DID problem tonight. Need to have Asterisk 1.0.3 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tim Lewis Sent: Tuesday, January 25, 2005 6:33 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Anyone h

Re: [Asterisk-Users] Re: bellster credits problem coming...

If I have * play a message telling the caller that the messages are logged and recorded as a condition of the use of my line, if they do not agree to this hang up. Would it be illegal to record then? dhh > This is why you need to make sure to keep your call detail records > around. Though it w

Re: [Asterisk-Users] size and quality of audio clips effect theplayback??

Have you compiled a vanilla kernel yet? I don't trust any distro supplied kernel. On Tue, 2005-01-25 at 12:07 -0800, Gabriel Afana wrote: > Anybody have any ideas on this? I dont know what to do and my new website > just launched yesteryday. > > www.gafana.com > > Go to the "Real-time sport sco

[Asterisk-Users] BroadVoice Help

Is the Broadvoice service up? I just signed up with them and started receiving calls in no time but could not make calls. And after a few minutes I cannot even place calls.   register => [number]:[EMAIL PROTECTED]     [broadvoice] type=peer fromuser=[number] host=proxy.lax.broadvoic

[Asterisk-Users] Interesting bellster issue

This morning I set up my bellster account and purposly drained the account to zero, so I could see how my * box will route the call. When I make a call, bellster anounces that I have no credits and says goodbye, but it still routes the call. Any ideas? I think this is a great idea. My friends a

[Asterisk-Users] Re: I think your problem has to do with how you set the variable.

No , forget the undescore I wrote an error in the email. All the variables prints out correctly in chiamamezzi-Wave context only after the Originate command succesfully run the dial application on chiamamezzi-dialout , _X.,1 This is a row flow of what happens: User Vars visible(of course

Re: [Asterisk-Users] R2 in Bolivia

Hi Jorge, You might be the first person to try the Bolivian variant. I need more information to make any sense of the problem. In /etc/asterisk/unicall.conf add the line: loglevel = 1023 and try again. You should get a much more detailed log of what happens. Send that to me. Regards, Steve [EM

[Asterisk-Users] Re: [Asterisk-biz] bellster.net - GREAT advance

Sam> In France, the second most important ADSL provider (named "Free") Sam> offers a phone line (which uses VoIP but can only be used as a FXS) Sam> with unlimited free calls to landlines. I also have Free ADSL in Paris, and would very much like to get their VoIP working natively with Asterisk. F

RE: [Asterisk-Users] Asterisk HEAD ->> Stable schedule?

[EMAIL PROTECTED] wrote: > Jim Van Meggelen wrote: > >> So 1.0.x STABLE will become 1.2.0 STABLE, and 1.1.x HEAD will be >> continued as 1.3.0 HEAD >> >> Could be wrong, but it'd make sense. > > It would make sense if we ever did unstable releases from HEAD, but we > don't do that currently :-)

Re: [Asterisk-Users] AMP with SUSE 9.2

One thing for sure it was a rela headache for me... I finaly did get it working... Don't for get to follow the installation guide to a "t"... Charles On Tue, 25 Jan 2005, Keith Burns wrote: > Hi, > > I have the newbie guide from AMP's website and (fair enough) it is all > about whitebox linux. H

[Asterisk-Users] E100P vs TE110P & Echo

Does anyone have any experience of having problems with E100P echo on a UK BT PRI and the improvements offered by the TE110P. I have a customer with significant echo problems using a E100P and I want to be sure that the TE110p will fix the issues before I invest in the new card. Regards, Stuart

RE: [Asterisk-Users] Re: bellster credits problem coming...

Not true -- even if you didn't do anything, but allowed someone else to use your equipment to do something, you can be held responsible -- and at the very least, you'll have a major mess on your hands. Personally, I consider my privacy (and that of my phone-line) to be more important than a few dol

Re: [Asterisk-Users] PrivacyManager not Working

Brian Dingman wrote: Keith, VP Connect is having issues right now with callerid being transmitted... as much as they don't want to believe it. Sometimes it works, sometimes it doesn't. Maybe this is part of the problem. Does PM not work 100% of the time for you? On Mon, 24 Jan 2005 21:29:37 -0500,

[Asterisk-Users] Anyone having problems with LiveVoIP?

I am having two problems. The first one is about half the time asterisk fails to read the DTMF tones. The second is with my 3 DID's some times it goes through and other times it does now. Right now it does nothing. Sometimes it rings for ever. With no out put on the asterisk console. They don't l

Re: [Asterisk-Users] New native assisted transfer (atxfer) usage info required

I had it working. My features.conf file is the same as yours except for the [featuremap]. I use "##" for blindxfer and "**" for atxfer. In my dial plan I use "t" or "T" as the Dial() flag. Make sure that you have beep.gsm and beeperr.gsm in the asterisk sound file folder. If these files are mi

[Asterisk-Users] grandstream budgetone-100 updates

I’m using tftp server that automatically loads on each reboot, for some reason the last 2 files fail to load each time. (and I think this has always been the case)     Aborted     192.168.16.32    C:\Program Files\TFTP Desktop\1.0.5.18\cfg000b82005c24   Octet, Send   

Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone

Robert Rozman wrote: - Original Message - From: "Steve Kann" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, January 25, 2005 3:56 PM Subject: Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone Robert Rozman wrote:

RE: [Asterisk-Users] Florz patch for zaphfc

Nils, Thanks for your help with this issue and I thought I should send this to the list for the benefit of others. The issue was with the options piping through "patch". The command I used was "zcat /path/zaphfc_0.2.0-rc2b_florz-1.diff.gz | patch" which worked fine. Notice no p1 option. The fee

RE: [Asterisk-Users] UPS for Asterisk

On Tue, 25 Jan 2005, David Brodbeck wrote: > > From: Peter Svensson [mailto:[EMAIL PROTECTED] > > The SmartUPS ups's from APC that are >= 1kVA seem to be of a > > lot better > > quality then their smaller siblings. We have lost none of the 1kVA or > > larger ups:es while several of the smaller

RE: [Asterisk-Users] Codec negotiation

I don't want that... because - for outbound calls I want priority to be g729 first - for inbound calls I want no priority at all (e.g. the calling asterisk to decide which codec we will use) The last doesn't happen.. This by the way DID happen correctly with previous versions of asterisk (1.0.3

[Asterisk-Users] Polycom and call waiting again..

I searched and read all the relevant posts, but I still don't have a solution to my problem.. I've got a small queue for tech support calls using AddQueueMember. The agents are using IP300's from polycom. In my example, only one agent is logged int. When a call comes into the queue, asterisk

RE: [Asterisk-Users] BroadVoice Or VoicePulse ?

My current favorites are www.iax.cc for DIDs and www.voipjet.com or www.simpletelecom.com for outgoing calls. I use multiple providers for outgoing calls based on rate and services; for example, simpletelecom allows for free 800# calls and lower Germany/Cell rates than voipjet; voipjet has 1.3c/mi

Re: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server

Has anyone had any luck with this issue and new Asterisk/Zaptel releases (1.05/1.04)? I am still searching for a solution and waiting for that Eureka! moment.. On Thu, 20 Jan 2005 09:20:09 +0100, Tais M. Hansen <[EMAIL PROTECTED]> wrote: > On Wednesday 19 January 2005 23:15, Eric Bishop wrote

Re: [Asterisk-Users] TDM400P Dell 1850 Server

On Jan 25, 2005, at 1:22 PM, Eric Wieling wrote: Adam Robins wrote: The TE410P is a T1/E1 card. I need the card for POTS lines. Is there also a TDM410P that does not appear on the Digium web site? The TDM400P only works in standard PCI 2.2. Not PCI-X, not PCI-Extreme, not PCI-64bit. Huh? It ob

[Asterisk-Users] Server side three-way calling with SIP channel

I have a SIP phone that doesn't support three-way calling. Is there a way to do three-way calling from a SIP phone server side instead? TKS __ Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do more. Manage less. http://info.mail.yahoo.com/ma

Re: [Asterisk-Users] Cisco 7905 automagically sends to VM (again...)

Hello Adi! Well that helps not to go to VM if called party doesn't pick up. But it doesn't if called party is busy. BR, Alen Adi Linden wrote: I put this into the configuration file for the 7905G. Solved all my forwarding to voicemail issues. # Some other defaults ForwardToVMDelay:4294967

Re: [Asterisk-Users] Re: bellster credits problem coming...

This is why you need to make sure to keep your call detail records around. Though it would likely be illegal, you could set up your asterisk box to record every conversation that went through it also. Bellster hopefully has logs of where calls originated from also. If you didn't do anything,

Re: [Asterisk-Users] Unable to Specify Channel 1 - no such device or address

Direct from the source (1.0.x) Here is a list of what module to use with what hardware Module Name Hardware tor2T400P - Quad Span T1 Card E400P - Quad Span E1 Card wct4xxp TE405P - Quad Span T1/E1 Card (5v version) TE410P - Quad Span T1/E1 Car

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