[asterisk-users] anyone remembers where to check this list threads on a web site?

Hi all, excuse this doll question, but can´t remember or find where I used to check this list on the web, email is becoming unmanageable along with my regular mail. can anyone provide me withe the link to check the list´s threads under web?:-[

Re: [asterisk-users] Best phone for easy provisioning

Best and easiest provisioning I´ve found imho is Snom, great web interfase , followed by Polycom (web interfase used to be poor and slow, but once you set it up, works very well) Dovid B escribió: I liked polycom a lot. - Original Message - *From:* Rod Bacon

[asterisk-users] does any one knows of a Softphone that works under terminal services?

Hi all I'm looking for a softphone that works well under terminal services environment, we need to set up 24 to 32 phones for a call center, also, does any one knows if it will actually work fine under load? ___ --Bandwidth and Colocation provid

Re: [asterisk-users] CTI: put on hold a call

I´m looking for the same feature performed with the manager, but I think should be the same problem you are experiencing I need to place music on hold (park) an specific call, while the agent performs a process/question/inquiry, and then retakes the call. Is there not a way to park the call?

Re: [asterisk-users] mwi for voicemail not showing up for realtime config.

Since I started using 1.4 I'm also not getting MWI. I am not using realtime. MARK. Benjamin Jacob wrote: Hello ppl, Am using realtime odbc storage for voicemail, sip users/peers, static for extensions and so on. My issue is I am not getting MWI for any fones, even tho I've got rtcachefriend

Re: [asterisk-users] how can I detect a DTMF tone while on a bridged call ? anyone knows?

ee features.conf for more details. On 10/10/06, MF <[EMAIL PROTECTED]> wrote: Hi all I have a call that comes in via a first E1 and goes out via a second one, my problem is I need to catch a digit dialed by the second one, during the conversation, I know I can detect the # for a call transfe

[asterisk-users] how can I detect a DTMF tone while on a bridged call ? anyone knows?

Hi all I have a call that comes in via a first E1 and goes out via a second one, my problem is I need to catch a digit dialed by the second one, during the conversation, I know I can detect the # for a call transfer, but nothing else, Any Idea on how to do this I deeply appreciate, !!

[asterisk-users] Is there a way to collect dtmf digits during a call? (inband)

Hi, Does anyone knows if I can collect DTMF digits (inband) during a bridged call (E1 to E1),? I see the DTMF tones on the debug file but does not activate the dialplan. My problem is, I need to signal from the second E1 to bridge the call to another E1 (a third one), if I use the tra

[asterisk-users] we are having trouble detecting the # for making a transfer from an E1, usually under some load, please help

Hi, this is the setting: a call comes in vi a first span g0,it get's hairpined to a second span g1, fine, on that second span we are accesing a legacy IVR,at some stage we need to get the call and transfer it (hairpin) to another number via the same first span g0, for that we di

[asterisk-users] Any one knows if I can connect SS7 to Asterisk, (with a TE405P or other) ???

Hi I need to connect at least 2 (and 2 more in the future) links to a switch via SS7, does anyone knows if this can be done with Digium cards? if not, which box could I use to convert from SS7 to isdn, (could anyone please recomend one of these boxes, and let me know a ballpark price fi

Re: [asterisk-users] I need to record 30 conversations and have other 30 with music on hold, all at the same time, can a PIV handle it?

drive (prior to this having mapped the directory where the files are, probably with NFS) . Steve Totaro escribió: Raphaël Jacquot wrote: MF wrote: Hi, I have a 2 E1 system with 32 zap FXS extensions (all Zaptel, with TDM2400), on a PIV, 3GHz, 1GB, Well my question is wether I'

Re: [asterisk-users] I need to record 30 conversations and have other 30 with music on hold, all at the same time, can a PIV handle it?

when fully loaded? What should I expect while working with PIV before switching to amd64?. thanks Manrique Raphaël Jacquot escribió: MF wrote: Hi, I have a 2 E1 system with 32 zap FXS extensions (all Zaptel, with TDM2400), on a PIV, 3GHz, 1GB, Well my question is wether I'll be a

[asterisk-users] I need to record 30 conversations and have other 30 with music on hold, all at the same time, can a PIV handle it?

Hi, I have a 2 E1 system with 32 zap FXS extensions (all Zaptel, with TDM2400), on a PIV, 3GHz, 1GB, Well my question is wether I'll be able to use it for peak demand moment, that is having all 60 channels busy 30 talking to agents on the FXS, while recording their conversation at the s

Re: [asterisk-users] How to notify an ACD agent before he/she picks up

Has anyone got a clue about this?I need to know which operator to send a message to, prior to the queue command ringing him, (just after he is assigned) Anyone knows if I can get to know the operator ACD choosed to send the call by using Realtime Queue, or maybe via the manager? Hi,

Re: [asterisk-users] Has anyone tried to install both digital card and analog card in one machine

We are currently working with two TDM2400P, with 32 FXS ports and one TE205P all on the same machine, and it works fine, (haven't done any stress testing yet though, maybe if someone could share his/her experience with high load) Xue Liangliang escribió: Hi, all, I am not sure whether we

Re: [asterisk-users] How to notify an ACD agent before he/she picks up

Anyone knows if I can get to know the operator ACD choosed to send the call by using Realtime Queue, or maybe via the manager? Hi, I need to send a message to an agent when the ACD starts to ring on he/she. I have and application already built that sends such a message (just like a cti),

[asterisk-users] I can´t set to work two tdm2400p and one TE205p on same machine, please help

Hi, I´m trying to set up this three boards, two tdm240p (with 32 FXS ports) and one TE205p, I appreciate if anyone could take a look at my config files to see what´s wrong. If I set only the TDM240p or only the TE205p they work fine separately, just can´t make them work together,

Re: [Asterisk-Users] MOH during M() macro execution

Did you try mM()? extensions.conf: exten => s,n(dial),Dial(SIP/sipura2_1&SIP/sipura2_2|30|mtTM(screen)) MARK. Ed Greenberg wrote: If I dial with an M() option to run a macro after answer, (how) can I get music on hold for the caller? ___ --Band

Re: [Asterisk-Users] TDM400P Dialing Out - "Cannot be completed as dialed"

I can't see anything immediately wrong here. Maybe it's something in your dialplan? It sounds like Asterisk is dialing out but the PSTN doesn't like the number you are dialing. Are you in an area that requires 10 digits or does not like if you dial 11 for local calls? From what I can tell,

Re: [Asterisk-Users] CallerID Name in dialplan

Set(CALLERID(name)=VOICE MAIL) Andy Vega wrote: Is it possible to show CallerID names for dialplan applications? When I call from phone-to-phone, it shows the CallerID from sip.conf or iax.conf, but I don't know of any way to show CallerID Name when I call the extension for an application (vo

Re: [Asterisk-Users] Question on Zap interfaces

Specify a different context for each Zap channel (context=homephone) (context=workline) in zapata.conf instead of just inbound-analog. Then in your extensions define a context for each that includes a different dialplan. On the second problem, you could remove the forward from the Verizon l

[Asterisk-Users] Qualify time +2000ms?

Although I'm convinced that Broadvoice doesn't have the most stable of ping times, it seems like I get ping results that are approximately the ping time +2000ms at times. Has anyone experienced this problem with qualify on a SIP connection before? So here, was the ping 20ms or 2020ms as repor

Re: [Asterisk-Users] Does anyone run Asterisk on FC4? with Digium's TDM40B cards

is Intel based. and there is no USB conflicts at all. I ran FC3 well, but, I think new kernel (in the FC4) may be the place to see. Thank you Kumara - Original Message - From: "MF Hulber" <[EMAIL PROTECTED]> To: "Kumara Jayaweera" <[EMAIL PROTECTED]>;

Re: [Asterisk-Users] Does anyone run Asterisk on FC4? with Digium's TDM40B cards

t for the reply. my box is Intel based. and there is no USB conflicts at all. I ran FC3 well, but, I think new kernel (in the FC4) may be the place to see. Thank you Kumara - Original Message - From: "MF Hulber" <[EMAIL PROTECTED]> To: "Kumara Jayaweera" <[

Re: [Asterisk-Users] Phone interface hardware

Without having said why you want to connect the phone through the computers I have a hard time understanding why you want to do that. As someone else has suggested, either buy IP phones and connect them directly to your LAN or buy analog adapters and use your existing phones. I don't see the

Re: [Asterisk-Users] Does anyone run Asterisk on FC4? with Digium's TDM40B cards

I don't use Fedora but I do use RHEL AS 4 without any problem. Do you have any USB conflicts? MARK. Kumara Jayaweera wrote: Hi all, Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4). Please any c

Re: [Asterisk-Users] low sound

It sounds like you need to adjust your txgain or possibly rxgain. MARK. jonny hashem wrote: my customers complain that when they make a call they hear the another side very well but the another side hears the first side well but in low sound.what is the ptoblem here and i have to change? __

Re: [Asterisk-Users] Zaptel warning

The next question is, "was your call successful?" I see you dialed an 8 digit number. Is that what's required on your line? MARK. Eric Wieling aka ManxPower wrote: VoIP Newbie wrote: Hi all, When I was making calls from an IP phone, through a X100P, to PSTN, the following error was encou

Re: [Asterisk-Users] Getting asterisk to work with callthroughs?

Ok, first I'll tell you some of the things I'm ignoring because you said you are having trouble receiving the inbound call. First, why aren't you using DISA? Ok, so you want to try this out, that's fine. Second, it appears you set the variable NR to be empty so I don't know why you are using

Re: [Asterisk-Users] Is this maillist down?

It's not just him. The list was majorly down from sometime on the 29th until the 1st. MARK. Derek Whitten wrote: must be just you.. get messages all day every day here.. :-) On Mon, 2005-08-01 at 05:49, Howard Leadmon wrote: This is usually a very active list, but looking at

Re: [Asterisk-Users] Gmail and the list

I don't think anyone was getting mail from any of the lists (or at least a very few were). I know that I wasn't but I have started to receive it now. MARK. Michel Koenen wrote: Same here, nothing is coming in anymore on my gmail address neither. I read your posting by going to the web versi

Re: [Asterisk-Users] Klicking sounds in background

Just a thought: do you have DSL on the PSTN line and are you using a line filter? MARK. Jochen Witte wrote: Hello, I have an IVR with Intel HMP SIP stack, which is a peer behind my Asterisk box (Asterisk 1.0.7, Digium PRI). When dialing in via PSTN, there are klicking sounds in the backgrou

Re: [Asterisk-Users] mpg123 - two processes

Yes, I always have two. MARK. Billy Dunn wrote: Does everyone have two processes running for mpg123? I always have them when I'm running an idle Asterisk box. No calls going in or out and nothing off hook. Is this normal? Thanks! 5008 ?S 0:00 mpg123 -q -s --mono -r 8000 -b

Re: [Asterisk-Users] NoOp

And is there some bit of information I get at verbose level 255 that I don't get at 254? It just seems like a lot of levels. MARK. John Novack wrote: MF Hulber wrote: It's a little odd. Something like "asterisk -v4" seems more appropriate. You can also use >s

Re: [Asterisk-Users] NoOp

eorge Garvey wrote: On Sun, Jul 10, 2005 at 09:49:37PM -0400, MF Hulber wrote: Maybe it shows up after a certain verbosity level. Try >asterisk -r When I do that NoOps always show up. Looks like you're right. Guess I never used enough v's ;) _

Re: [Asterisk-Users] NoOp

Maybe it shows up after a certain verbosity level. Try >asterisk -r When I do that NoOps always show up. MARK. George Garvey wrote: I believed from reading that NoOp would display something on the console. I assume the console is * in the foreground. During testing, I've often been runnin

Re: [Asterisk-Users] Definitive CallerID Format and anonymous?

The new format is: exten => _1NX,1,Set(CALLERID(number)=4025551212|a) exten => _1NX,2,Set(CALLERID(name)=NPI|a) exten => _1NX,3,Dial(IAX2/teliaxout/${EXTEN}) MARK. Rich Adamson wrote: Thanks for the thorough reply. I'm aware that ther

Re: [Asterisk-Users] Newbie Question: Type of card

Take a look here: http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P MARK. Dan Adams wrote: Hi, I am sorta a newbie to the asterisk community at least in the realm of hardware types. I was wondering, what type of card is used to allow asterisk, on a slackwa

[Asterisk-Users] Bounced mail apologies

My apologies for any bounced mail from me today. My mail server was having a bit of a fit. MARK. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options v

Re: [Asterisk-Users] Sip.conf problems

Try two different entries: sip.conf: [general] ;disallow=gsm ;allow=ulaw port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls callerid=No CallID register => user

[Asterisk-Users] Cell Phones reporting internation calls

I have two different inbound DID providers and on each network I have had reports in the last several days from callers that when they try to call me from a cell phone their provider reports that they are trying to make an international call and thus doesn't let them complete the calls. In eac

[Asterisk-Users] UTStarCom F1000 SIP configuration

Has anyone had any luck configuring the UTStarCom F1000 with asterisk? I get the wireless to work but the sip registration is a problem. Below is my SIP Debug. The server is 192.168.0.80 and the phone is 192.168.0.166. Sip.conf: [f1000_1] type=friend host=dynamic defaultip=192.168.

Re: [Asterisk-Users] Inbound provider in Canada

I have Canada DIDs from the companies below and both have been reliable: http://www.unlimitel.ca http://www.livevoip.com MARK. Adrian A wrote: Does anyone have any recommendations for a SIP/IAX provider I can use for inbound callls? The plan is to have a 1800 number people can call and

Re: [Asterisk-Users] OT: Please comment on Dvorak's troll

Unfortunately I believe there is a lot of truth to it. The speed in which 911 legislation took effect is no coincidence and you don't see the big telcos complaining about it. He's right about the price issue too. Do you see how much big providers charge for VoIP service? MARK. Colin Anders

[Asterisk-Users] Degraded voice without packet loss

In making calls outbound through various VoIP origination providers people on the PSTN network have been complaining of poor voice quality. Diagnosing has been difficult with limited feedback but I tried using different originators and seem to have the same problem. I used TestYourVoip.com an

Re: [Asterisk-Users] Size of extensions.conf

Although you may not see it displayed on a reload it may actually be loaded. Try "show dialplan" and its alternatives to be sure that what you are looking for is not loaded. MARK. John Melody wrote: Is there a limit to the number of extensions that can be defined in extensions.conf ? I just

Re: [Asterisk-Users] livevoip

I can't disagree with you on the customer service aspects. I have found the new online ticket reporting system a bit better but I often get the feeling that they feel customers are a nuisance. As far as the VoIP service, I haven't had any problems to speak of. I'm still waiting for a DID tha

[Asterisk-Users] New IAXy available?

Anyone know when the new IAXy will be available and what changes there are besides the form factor? MARK. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update op

Re: [Asterisk-Users] MOH per User

I've tried it by changing the MOH context after I have identified the caller but I find that mpg123 doesn't properly switch over to the new context. Once it starts playing a stream it appears to be stuck with it. exten => s,1,SetMusicOnHold(default) MARK. [EMAIL PROTECTED] wrote: Is it possib

[Asterisk-Users] Linksys/Cisco buys Sipura

Have you seen this story? Cisco definitely wants to own the VoIP market. I wonder what effect this will have on Sipura products. http://story.news.yahoo.com/news?tmpl=story&u=/nf/20050427/bs_nf/33554 MARK. ___ Asterisk-Users mailing list Asterisk-User

Re: [Asterisk-Users] Re: QOS Routers

These things are dirt cheap. Are they any good? MARK. Iassen Hristov wrote: Maybe this fits the bill. It retails for less than $100 Message: 9 Date: Fri, 22 Apr 2005 10:42:20 -0700 From: Max Clark <[EMAIL PROTECTED]> Subject: [Aste

Re: [Asterisk-Users] livevoip callerid

I don't think it's correct to put dashes in the CIDNum. MARK. Paul Fielding wrote: Hmmm... I still can't get name, though number works. Perhaps I'm missing something? context livevoip in iax.conf that hooks me to livevoip dial 9 in front of long distance number to dial livevoip instead of regul

Re: [Asterisk-Users] GotoIf in Stable 1.0.4

Try terminating the GotoIf statement with a ')' MARK. Mark Halverson wrote: I have unlimited local calling on my cell phone provider but not long distance; so I wanted to create authentication based on me calling in and authenticating based on the callerid of my cell phone. Here is what I tried bas

Re: [Asterisk-Users] BroadVoice Inbound Not working ..

BTW, you hijacked someone's thread... Try this: -- sip.conf -- register => 5595551212:[EMAIL PROTECTED]/5595551212 [broadvoice-in] type=peer host=sip.broadvoice.com context=broadvoice-inbound canreinvite=no qualify=yes nat=no -- extensions.conf -- [broadvoice-inbound] exten => 5595551212,1,Goto(d

Re: [Asterisk-Users] Queue works, but the caller hears silence instead of ring tone

I believe there is a problem with LiveVoip ringback that is in the process of being resolved. MARK. Jon Califf wrote: If I call my 800 number, I do correctly go to the queue, the agents are dialed and the call can be picked up but I don't hear any ring tone while I'm waiting. If I pick up one o

Re: [Asterisk-Users] asterisk sounds

asterisk/sounds.txt Josiah Bryan wrote: On Tuesday 05 April 2005 1:24 pm, Dov Bigio wrote: Hello all, I am looking for a list of all available sound files for asterisk and a transcription of their content, so that I can have someone translate them into portuguese. I vaguely remeber reading

Re: [Asterisk-Users] livevoip callerid

I am able to set name and number with Livevoip. Make sure your variables are actually being set. exten => s,1,SetCIDNum(xx|a) exten => s,n,SetCIDName(first last|a) exten => s,n,Noop(Caller Name: ${CALLERIDNAME}, Number: ${CALLERIDNUM}) MARK. Cameron Schaus wrote: Is there any way I

Re: [Asterisk-Users] Looping messages

You might try adding: exten => h,1,Hangup Chris Blake wrote: Greetings *`s, I have set up a call which constantly loops a pre-recorded message waiting for the user to press a digit on their phone. At this point the call is sent elsewhere in the dialplan. But if the called party doesn`t press any bu

Re: [Asterisk-Users] Problem with livevoip dial out

iax.conf [livevoip-out] type=peer host=217.160.244.186 auth=md5 context=livevoip-dialout callerid="aaa bbb" <(xxx) xxx-> username=username secret=secret qualify=yes notransfer=yes exten => s,n,SetVar(LIVEVOIP=IAX2/[EMAIL PROTECTED]) exten => s,n(dial1),Dial(${LIVEVOIP}/dialed-numbe

Re: [Asterisk-Users] sip.conf match

The way it works with my provider is that although both numbers enter the same context, each number will match its own extension. If I have two numbers: 11 and 22 it works as follows: [sip-in] exten => 11,1,Noop(First number dialed) exten => 22,1,Noop(Second numb

Re: [Asterisk-Users] auto navigate external IVR

asterisk*CLI> show application dial 'D([digits])' -- Send DTMF digit string *after* called party has answered but before the bridge. (w=500ms sec pause) MARK. Shadow Roldan wrote: Hi Everyone I'm trying to pass a call to a outbound SIP peer(broadvoice) and pass DTMF tones afte

Re: [Asterisk-Users] Livevoip still no DTMF?

I don't have any difficulty with DTMF with LiveVoip incoming or outgoing. MARK. Brian Litzinger wrote: I read in the archives a number of discussions about livevoip, DID, and DTMF not working. However, no resolutions. I just setup a livevoip DID and indeed the DTMF does not work. The same asterisk

Re: [Asterisk-Users] Wiki down?

Me too. MARK. Tim Connolly wrote: Anyone else seeing problems trying to browse the wiki? Like no response on port 80? ___ Asterisk-Users mailing list Asterisk-Users@lis

Re: [Asterisk-Users] Broadvoice getting unregistered

For fun, try changing your context from [sip.broadvoice.com] to your phone number [55] MARK. Courtney Couch wrote: The asterisk config that i have is: [sip.broadvoice.com] type=peer authname=55 canreinvite=no context=test dtmf=inband dtmfmode=inband fromdomain=sip.broadvoice.com f

Re: [Asterisk-Users] voicemail sending blank .WAV file via email

voicemail.conf ;format=gsm|wav49|wav ;format=wav|wav49|gsm format=wav Jim Sturtevant wrote: I’ve recently installed asterisk and am working with the email a voicemail function. When a voice msg is left 4 files are created in the /var/spool… directory. They are .gsm, .txt, .wav and .WAV. The .wav

Re: [Asterisk-Users] How to do something random?

Take a look at the Random() command. MARK. Ronald Wiplinger wrote: I want to change the below lines: exten => _011.,1,SetGroup(line1); set current group to line exten => _011.,2,CheckGroup(1); check line1 does not have more than 1 exten => _011.,3,Dial,SIP/[EMA

Re: [Asterisk-Users] DTMF tones not working

Is it correct to have the same context (202) listed twice in sip.conf? Courtney Couch wrote: I have Polycom ip-300 phones that worked yesterday but dont seem to work today (at least dtmf signalling once connected to the asterisk box) The current configuration is: [general] port = 5060 bindaddr =

Re: [Asterisk-Users] uniden voip gear

Haven't used it since I don't think this is available yet. The only thing that bothers me about this unit is that it's single line. I currently have a similar analog Panasonic 2.4GHz system. In general I like the Panasonic except fir a couple things: They never seem to support the mute funct

Re: [Asterisk-Users] voicemail problems with CVS-HEAD

All of my sounds are under /var/lib/asterisk/sounds. I don't have a directory /usr/share/asterisk. None of my configuration files have a pointer to a sounds directory so I'm assuming it's looking in /var/lib/asterisk/sounds by default. MARK. G.Marshall wrote: Hello, I have moved from Asterisk

Re: [Asterisk-Users] SIP response *

rfc3261 http://www.faqs.org/rfcs/rfc3261.html Ronald Wiplinger wrote: Where can I get a list of all possible SIP ... response numbers and their meaning? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/

Re: [Asterisk-Users] :: BIOS Motherboard Settings ::

I have the same motherboard. I put the card in the 2nd slot from the bottom. In this slot, if you look at the manual, it will possibly be in conflict with some USB channels. I believe I may have disabled one of them but in any event I'm not using any USB devices. Otherwise, I didn't have to

Re: [Asterisk-Users] Script to Authenticate User and Dial Out

It seems the simplest approach is to create an extension(s) with the password(s) and then if the incoming caller gets it correct, jump them to another context. Otherwise stay in the default context and give default prompts. I suppose you can find a way to read the password from a DB or file o

Re: [Asterisk-Users] Log Error

I'm getting the same problem running Asterisk CVS-HEAD-03/21/05-03:24:01 built by [EMAIL PROTECTED] on a i686 running Linux which is the code from yesterday. Robert Goodyear wrote: FWIW I get the same exact error at the end of every VM session as well, thus: -- Playing 'vm-intro' (language

Re: [Asterisk-Users] VoIP service through Asterisk?

I took a look at teliax.  The pay as you go plan appears not to include international dialing and the commercial plan is fixed price of $44.99 per month capped at 500 international minutes a month.  Are you aware if they have international rates based on usage? MARK. Rich Adamson wrote:

Re: [Asterisk-Users] VoIP service through Asterisk?

Livevoip.com is another one that's pretty reliable and only charge 1.27 cents per minute. You can add a toll-free number for an extra $1 per month. Setup was a breeze! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of MF Hulber Sent: Sunday, March 20,

Re: [Asterisk-Users] VoIP service through Asterisk?

I started with Broadvoice recently but I am constantly having problems. I got everything configured and now it is dropping outgoing calls after 40 seconds and incoming calls are going direct to voicemail. Getting customer service is proving very difficult. I've started to look at some of the

Re: [Asterisk-Users] Broadvoice hangs-up / disconnects after about 30 deconds

I have the same problem but not with X-Lite.  I was using Broadvoice all day today and then I changed rate plans because I thought everything was working well.  Now my calls get dropped within 2 minutes and my incoming calls go direct to broadvoice voicemail. MARK. Scott Wolfe wrote:

Re: ***SOLVED*** [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server ****Solved*****!

I concur. I rebuilt today and now I seem to be able to dial out. MARK. Chris Nibeck wrote: thank you everyone! It does not seen that it was configuration problems at all. It appears it was the CVS that I was using from yesterday. I decided to start over, downloaded the latest CVS, recompiled, and

Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server

Try changing the extension from Broadvoice1 to the actual phone number (and don't send your secret in a public email or maybe that's Chris'): [*8475100139*] type=peer ;user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=8475100139 secret=XXX username=8475100139 Zanz

[Fwd: Re: [Asterisk-Users] BroadVoice configuration changes for Outbound]

Original Message Subject: Re: [Asterisk-Users] BroadVoice configuration changes for Outbound Date: Sun, 06 Mar 2005 19:11:22 -0500 From: MF Hulber <[EMAIL PROTECTED]> To: As

Re: [Asterisk-Users] MusicOnHold

Sorry, in my haste I didn't read your musiconhold.conf that answers my question about setting up the executable. MARK. Ken Godee wrote: MF Hulber wrote: I'm looking for a simple way to disable MusicOnHold in my environment. I'm not really interested in having it and it c

Re: [Asterisk-Users] MusicOnHold

I'll give that a try. Do you make a symbolic link to Madplay as mpg123 or is there a way to configure * to use a different executable? An issue I have with MOH at all is that if I'm on a conference call, I don't want MOH to play. MARK. Ken Godee wrote: MF Hulber wrote: I&

[Asterisk-Users] MusicOnHold

I'm looking for a simple way to disable MusicOnHold in my environment. I'm not really interested in having it and it causes too many problems with hanging mpg123 processes and memory management errors. The problem is, so many other modules seem to depend on it. I can't just cause a noload of