Re: [asterisk-users] IF else
On Wed, 19 Nov 2008, michel freiha wrote:
Hi all,
I have the following context in extensions.conf:
[a2billing]
exten = _X.,1,Gotoif($[${EXTEN} = 111] ? 21)
exten = _X.,2,DeadAGI,a2billing.php
exten = _X.,3,Wait,2
exten = _X.,4,Hangup
exten = _X.,21,Playback(AR_GetGiveToID)
exten = _X.,22,Wait(2)
exten = _X.,23,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _X.,24,Wait(2)
exten = _X.,25,Playback(/tmp/asterisk-recording)
exten = _X.,26,Wait(2)
exten = _X.,27,Hangup
If the customer dial 111, it'll be router to the entry with priority 21,
else it'll go to priority 2...I would like to add a third condition that if
the user dial let's say 112 it'll go to the priority 28 let's say
1. Stop using numbers.
2. Start using labels.
3. Add comments.
exten = _X.,1,Gotoif($[${EXTEN} = 111]?exten111)
exten = _X.,n,Gotoif($[${EXTEN} = 112]?exten112)
exten = _X.,n,Noop(Didn't dial 111 or 112)
exten = _X.,n,DeadAGI,a2billing.php
exten = _X.,n,Wait,2
exten = _X.,n,Hangup
exten = _X.,n(exten111),Noop(Dialled 111)
exten = _X.,n,Playback(AR_GetGiveToID)
exten = _X.,n,Wait(2)
exten = _X.,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _X.,n,Wait(2)
exten = _X.,n,Playback(/tmp/asterisk-recording)
exten = _X.,n,Wait(2)
exten = _X.,n,Hangup
exten = _X.,n(exten112),Noop(Dialed 112)
exten = _X.,n,Playback(AR_GetGiveToID)
exten = _X.,n,Wait(2)
exten = _X.,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _X.,n,Wait(2)
exten = _X.,n,Playback(/tmp/asterisk-recording)
exten = _X.,n,Wait(2)
exten = _X.,n,Hangup
Gordon
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Re: [asterisk-users] IF else
On Wed, Nov 19, 2008 at 4:05 PM, Gordon Henderson
[EMAIL PROTECTED] wrote:
On Wed, 19 Nov 2008, michel freiha wrote:
Hi all,
I have the following context in extensions.conf:
[a2billing]
exten = _X.,1,Gotoif($[${EXTEN} = 111] ? 21)
exten = _X.,2,DeadAGI,a2billing.php
exten = _X.,3,Wait,2
exten = _X.,4,Hangup
exten = _X.,21,Playback(AR_GetGiveToID)
exten = _X.,22,Wait(2)
exten = _X.,23,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _X.,24,Wait(2)
exten = _X.,25,Playback(/tmp/asterisk-recording)
exten = _X.,26,Wait(2)
exten = _X.,27,Hangup
If the customer dial 111, it'll be router to the entry with priority 21,
else it'll go to priority 2...I would like to add a third condition that if
the user dial let's say 112 it'll go to the priority 28 let's say
1. Stop using numbers.
2. Start using labels.
3. Add comments.
exten = _X.,1,Gotoif($[${EXTEN} = 111]?exten111)
exten = _X.,n,Gotoif($[${EXTEN} = 112]?exten112)
exten = _X.,n,Noop(Didn't dial 111 or 112)
exten = _X.,n,DeadAGI,a2billing.php
exten = _X.,n,Wait,2
exten = _X.,n,Hangup
exten = _X.,n(exten111),Noop(Dialled 111)
exten = _X.,n,Playback(AR_GetGiveToID)
exten = _X.,n,Wait(2)
exten = _X.,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _X.,n,Wait(2)
exten = _X.,n,Playback(/tmp/asterisk-recording)
exten = _X.,n,Wait(2)
exten = _X.,n,Hangup
exten = _X.,n(exten112),Noop(Dialed 112)
exten = _X.,n,Playback(AR_GetGiveToID)
exten = _X.,n,Wait(2)
exten = _X.,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _X.,n,Wait(2)
exten = _X.,n,Playback(/tmp/asterisk-recording)
exten = _X.,n,Wait(2)
exten = _X.,n,Hangup
1) Start using AEL (remove this context from extensions.conf and add
to extensions.ael):
context a2billing {
_X. = {
if(${EXTEN}=111) {
Playback(AR_GetGiveToID);
Wait(2);
Record(/tmp/asterisk-recording:ulaw,,5);
Wait(2);
Playback(/tmp/asterisk-recording);
Wait(2);
Hangup();
} else if(${EXTEN}=112) {
Playback(AR_GetGiveToID);
Wait(2);
Record(/tmp/asterisk-recording:ulaw,,5);
Wait(2);
Playback(/tmp/asterisk-recording);
Wait(2);
Hangup();
} else {
DeadAGI(a2billing.php);
Wait(2)
Hangup();
}
}
2) Start using extension masks (also works with AEL):
[a2billing]
exten = _111,1,Noop(Dialled 111)
exten = _111,n,Playback(AR_GetGiveToID)
exten = _111,n,Wait(2)
exten = _111,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _111,n,Wait(2)
exten = _111,n,Playback(/tmp/asterisk-recording)
exten = _111,n,Wait(2)
exten = _111,n,Hangup
exten = _112,1,Noop(Dialed 112)
exten = _112,n,Playback(AR_GetGiveToID)
exten = _112,n,Wait(2)
exten = _112,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _112,n,Wait(2)
exten = _112,n,Playback(/tmp/asterisk-recording)
exten = _112,n,Wait(2)
exten = _112,n,Hangup
exten = _X.,1,Noop(Didn't dial 111 or 112)
exten = _X.,n,DeadAGI,a2billing.php
exten = _X.,n,Wait,2
exten = _X.,n,Hangup
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] IF else
On Wed, 19 Nov 2008, Atis Lezdins wrote:
1) Start using AEL (remove this context from extensions.conf and add
to extensions.ael):
context a2billing {
_X. = {
if(${EXTEN}=111) {
Playback(AR_GetGiveToID);
Wait(2);
Record(/tmp/asterisk-recording:ulaw,,5);
Wait(2);
Playback(/tmp/asterisk-recording);
Wait(2);
Hangup();
} else if(${EXTEN}=112) {
Playback(AR_GetGiveToID);
Wait(2);
Record(/tmp/asterisk-recording:ulaw,,5);
Wait(2);
Playback(/tmp/asterisk-recording);
Wait(2);
Hangup();
} else {
DeadAGI(a2billing.php);
Wait(2)
Hangup();
}
}
You're missing a couple of semi-colons.
2) Start using extension masks (also works with AEL):
[a2billing]
exten = _111,1,Noop(Dialled 111)
exten = _111,n,Playback(AR_GetGiveToID)
exten = _111,n,Wait(2)
exten = _111,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _111,n,Wait(2)
exten = _111,n,Playback(/tmp/asterisk-recording)
exten = _111,n,Wait(2)
exten = _111,n,Hangup
exten = _112,1,Noop(Dialed 112)
exten = _112,n,Playback(AR_GetGiveToID)
exten = _112,n,Wait(2)
exten = _112,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _112,n,Wait(2)
exten = _112,n,Playback(/tmp/asterisk-recording)
exten = _112,n,Wait(2)
exten = _112,n,Hangup
exten = _X.,1,Noop(Didn't dial 111 or 112)
exten = _X.,n,DeadAGI,a2billing.php
exten = _X.,n,Wait,2
exten = _X.,n,Hangup
And, just in case the 2 extensions really are supposed to do the exact
same thing, use extension pattern matching:
context a2billing
{
_11[12] =
{
playback(AR_GetGiveToID);
wait(2);
record(/tmp/asterisk-recording:ulaw,,5);
wait(2);
playback(/tmp/asterisk-recording);
wait(2);
hangup();
};
_x. =
{
deadagi(a2billing.php);
wait(2);
hangup();
};
};
(The above is my first attempt at AEL. It parses, but it hasn't actually
been tested.)
I would question the use of deadagi() in a non-h extension. Are signals
not being trapped correctly in a2billing.php?
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] IF else
On Wed, Nov 19, 2008 at 6:51 PM, Steve Edwards
[EMAIL PROTECTED] wrote:
On Wed, 19 Nov 2008, Atis Lezdins wrote:
1) Start using AEL (remove this context from extensions.conf and add
to extensions.ael):
context a2billing {
_X. = {
if(${EXTEN}=111) {
Playback(AR_GetGiveToID);
Wait(2);
Record(/tmp/asterisk-recording:ulaw,,5);
Wait(2);
Playback(/tmp/asterisk-recording);
Wait(2);
Hangup();
} else if(${EXTEN}=112) {
Playback(AR_GetGiveToID);
Wait(2);
Record(/tmp/asterisk-recording:ulaw,,5);
Wait(2);
Playback(/tmp/asterisk-recording);
Wait(2);
Hangup();
} else {
DeadAGI(a2billing.php);
Wait(2)
Hangup();
}
}
You're missing a couple of semi-colons.
Sorry, that was untested proof of options :)
2) Start using extension masks (also works with AEL):
[a2billing]
exten = _111,1,Noop(Dialled 111)
exten = _111,n,Playback(AR_GetGiveToID)
exten = _111,n,Wait(2)
exten = _111,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _111,n,Wait(2)
exten = _111,n,Playback(/tmp/asterisk-recording)
exten = _111,n,Wait(2)
exten = _111,n,Hangup
exten = _112,1,Noop(Dialed 112)
exten = _112,n,Playback(AR_GetGiveToID)
exten = _112,n,Wait(2)
exten = _112,n,Record(/tmp/asterisk-recording:ulaw,,5)
exten = _112,n,Wait(2)
exten = _112,n,Playback(/tmp/asterisk-recording)
exten = _112,n,Wait(2)
exten = _112,n,Hangup
exten = _X.,1,Noop(Didn't dial 111 or 112)
exten = _X.,n,DeadAGI,a2billing.php
exten = _X.,n,Wait,2
exten = _X.,n,Hangup
And, just in case the 2 extensions really are supposed to do the exact
same thing, use extension pattern matching:
context a2billing
{
_11[12] =
{
playback(AR_GetGiveToID);
wait(2);
record(/tmp/asterisk-recording:ulaw,,5);
wait(2);
playback(/tmp/asterisk-recording);
wait(2);
hangup();
};
_x. =
{
deadagi(a2billing.php);
wait(2);
hangup();
};
};
(The above is my first attempt at AEL. It parses, but it hasn't actually
been tested.)
I would question the use of deadagi() in a non-h extension. Are signals
not being trapped correctly in a2billing.php?
AFAIK that's how a2billing is built, it's intentionally DeadAGI on
live channel. Ugly hack that gives warnings all the time in logs, but
it works and seems to provide correct billing info :)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] Anyone else having problems with the list
Yes for me. Carlos -- Julian Lyndon-Smith wrote: I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone else having problems with the list
On 22:43, Tue 25 Sep 07, Carlos Hernandez wrote: Yes for me. Carlos -- Julian Lyndon-Smith wrote: I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian I have similar problems. Some mails arrive, some dont. If I check the listarchive on the web I see more emails then in mutt. I already disabled greylisting etc and browsed thru the spam quarantine but nothing. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone else having problems with the list
me too :) Original-Nachricht Datum: Tue, 25 Sep 2007 12:57:25 +0200 Von: Michiel van Baak [EMAIL PROTECTED] An: asterisk-users@lists.digium.com Betreff: Re: [asterisk-users] Anyone else having problems with the list On 22:43, Tue 25 Sep 07, Carlos Hernandez wrote: Yes for me. Carlos -- Julian Lyndon-Smith wrote: I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). Is anyone else encountering this ? Julian I have similar problems. Some mails arrive, some dont. If I check the listarchive on the web I see more emails then in mutt. I already disabled greylisting etc and browsed thru the spam quarantine but nothing. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- GMX FreeMail: 1 GB Postfach, 5 E-Mail-Adressen, 10 Free SMS. Alle Infos und kostenlose Anmeldung: http://www.gmx.net/de/go/freemail ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone else having problems with the list
On Tue, 2007-09-25 at 10:14 +0100, Julian Lyndon-Smith wrote: I have sent a few emails over the past couple of days that simply have not arrived on the list (or so it seems). I'll take a look at this again... I thought we had most of the problems with the mailing lists fixed, but we seem to be having some problems again. (This is most likely our spam-catching system being over-aggressive, but I'll look into it.) -- Jared Smith Community Relations Manager Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
trixter http://www.0xdecafbad.com wrote: I am curious though about a companies competence when they have a production system and it takes a week of multiple outages to chnage something. You would think any professional company would have a test and development network seperate from the production one where they can *anounce* and schedule downtime for infrastructure changes. Those were my exact thoughts. Had I received a warning, I might have been able to do something about it.. like set call forwarding to go to my cell phone until they worked out their kinks. Unprofessional in my view. A response to their email, even an auto responder noting an outage in some area would keep me from getting so hot under the collar. All in all though my only complaint with broadvoice is that their tech support knows very little on average even about broadvoice specific things, like their rate plans and what is actually included with each package (ie which exchanges are within which subpackage for a given country). To bil this information has to exist somewhere, you would think that on a corporate level they would make this information more available like vonage does, but I can live without that. They do have enough information for most on their web site. The account portal seems put together enough. JD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
Andre Normandin wrote: Hello, About 4PM EDT I noticed that my broadvoice service cannot register.. Anyone else having problems with their broadvoice service? FYI: I connect to the 147.135.20.128 (nyc) proxy... Thanks, - Andre ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I'm down too. BROADVOICE do you watch this list? This is twice in seven days that you've had an outage. JD -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
I'm down too. BROADVOICE do you watch this list? This is twice in seven days that you've had an outage. JD My connection is back up. Maybe they DO read this list :) -- JD Austin Twin Geckos Technology Services LLC email: [EMAIL PROTECTED] http://www.twingeckos.com phone/fax: 480.422.1250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
Its like they are rebooting their sip proxies, my sip connection has been up/down all day, when it starts to come back it returns a 404 and then finally registers. I think this is related to the upgrades they have been doing for about a week now, which I also believe are related to the FCC ruling that VoIP providers in the US who connect to the PSTN provide CALEA (wiretap) support or be fined. On Mon, 2005-05-02 at 16:00 -0700, JD Austin wrote: Andre Normandin wrote: Hello, About 4PM EDT I noticed that my broadvoice service cannot register.. Anyone else having problems with their broadvoice service? FYI: I connect to the 147.135.20.128 (nyc) proxy... Thanks, - Andre I'm down too. BROADVOICE do you watch this list? This is twice in seven days that you've had an outage. JD -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else having Broadvoice issues today?
Probably a coincidence, but Teliax has been very choppy audio today. Most of my callers from 1 to 4 pm were Mr. Roboto. Please let it NOT be a trend... Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Jerry Geis |Sent: Monday, May 02, 2005 5:18 PM |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] Anyone else having Broadvoice issues today? | |yes | |from sometime after 4 to around 5:30 pm. | |jerry | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
I haven't lost my registration, but calls from Verizon, Nextel, and Sprint cell phones to my BV number (which worked in the past) are now getting a This call cannot be completed as dialed, please check the number and try again message. Calls from landlines seem to work fine, and my Rogers phone which is currently roaming on Cingular connects fine. Normally I would think that it was a routing issue (like people had with certain exchanges and Cingular in the past), but between three carriers from phones that have used this number daily for the past 2 months or so? Anyone else have this happen? I called BV support and they said they must be dialing the number wrong and didn't want to do anything even after I explained how that is not the case. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else having Broadvoice issues today?
BroadVoice is once again having issues .. There tech support is also reporting issues with no ETR so they don't even know what's wrong .. Inbound calls are hit and miss ... Outbound calls are DOA. BRW -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Gutowski Sent: Monday, May 02, 2005 7:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Anyone else having Broadvoice issues today? I haven't lost my registration, but calls from Verizon, Nextel, and Sprint cell phones to my BV number (which worked in the past) are now getting a This call cannot be completed as dialed, please check the number and try again message. Calls from landlines seem to work fine, and my Rogers phone which is currently roaming on Cingular connects fine. Normally I would think that it was a routing issue (like people had with certain exchanges and Cingular in the past), but between three carriers from phones that have used this number daily for the past 2 months or so? Anyone else have this happen? I called BV support and they said they must be dialing the number wrong and didn't want to do anything even after I explained how that is not the case. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else having Broadvoice issues today?
Mine seemed to have come back somewhere between 5:30 and 6... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jerry Geis Sent: Monday, May 02, 2005 8:18 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Anyone else having Broadvoice issues today? yes from sometime after 4 to around 5:30 pm. jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
Don't say that! i just moved from broadvoice to teliax!!! On 5/2/05, Andre Normandin [EMAIL PROTECTED] wrote: Mine seemed to have come back somewhere between 5:30 and 6... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jerry Geis Sent: Monday, May 02, 2005 8:18 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Anyone else having Broadvoice issues today? yes from sometime after 4 to around 5:30 pm. jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
Must be number dependent -- out of 12 numbers on one server, 2 regs failed (404 not found) for about 15 minutes around 4:30pm PST, but are fine since. Other numbers remained unaffected and work find incoming and outgoing... BTW, all numbers have the same area code and exchange. Interesting... --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else having Broadvoice issues today?
Boy.. with all this discussion about broadvoice lately, makes me appreciate my nufone account.. :-) On Mon, 2005-05-02 at 19:46, Andre Normandin wrote: Mine seemed to have come back somewhere between 5:30 and 6... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jerry Geis Sent: Monday, May 02, 2005 8:18 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Anyone else having Broadvoice issues today? yes from sometime after 4 to around 5:30 pm. jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Derek Whitten [EMAIL PROTECTED] kFuQ Productions signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else having Broadvoice issues today?
I've only had teliax for about 2 weeks or so, and was impressed at first, but their complete LACK of customer support is truly sad. I think it's a one-man operation and he has a second job or something. Never a callback until a threatening email, and maybe a few hours in the morning is all Ive seen someone on the 'java chat' tech window. I'm already looking for a replacement. (I wonder how broadvoice is? LOL) Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Mark Musone |Sent: Monday, May 02, 2005 8:12 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Anyone else having Broadvoice issues today? | |Don't say that! i just moved from broadvoice to teliax!!! | | | |On 5/2/05, Andre Normandin [EMAIL PROTECTED] wrote: | Mine seemed to have come back somewhere between 5:30 and 6... | | -Original Message- | From: [EMAIL PROTECTED] | [mailto:[EMAIL PROTECTED] Behalf Of Jerry Geis | Sent: Monday, May 02, 2005 8:18 PM | To: asterisk-users@lists.digium.com | Subject: [Asterisk-Users] Anyone else having Broadvoice issues today? | | yes | | from sometime after 4 to around 5:30 pm. | | jerry | | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
I wonder how broadvoice is? LOL Quite frankly, they still are on my personal recommended list. Last week's downtime (those few hours when nothing worked, remember?) was the first outage this year for me. And yes, I'm using those accounts actively, each has easily 10-15 calls a day. I don't call today an outage since 10 out of 12 numbers worked for me, and those two only has issues for 15 minutes. Stuff happens; I guess they are implementing the required wire-tapping and/or 911 stuff... or whatever. --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice issues today?
On Mon, 2005-05-02 at 21:39 -0700, Luki wrote: has issues for 15 minutes. Stuff happens; I guess they are implementing the required wire-tapping and/or 911 stuff... or whatever. I think its fair to say they are doing something :) I also have few problems with them in general, just stuff happened recently, and while for some things, temporary outages arent that big of a deal, for many telephone stuff is more critical to their business or whatever. VoIP in general is new, and the legislation for US based (or those that operate there) companies is changing which is going to force them to make infrastructure changes. Depending on how it was originally set up it depends on how severe that is. I am curious though about a companies competence when they have a production system and it takes a week of multiple outages to chnage something. You would think any professional company would have a test and development network seperate from the production one where they can *anounce* and schedule downtime for infrastructure changes. All in all though my only complaint with broadvoice is that their tech support knows very little on average even about broadvoice specific things, like their rate plans and what is actually included with each package (ie which exchanges are within which subpackage for a given country). To bil this information has to exist somewhere, you would think that on a corporate level they would make this information more available like vonage does, but I can live without that. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else having Broadvoice issues today?
No.. but... In their defense though, Cisco sold us a million dollars in routers taunting they could handle the load. 6 months later we were trading them in for Junipers because they were only able to handle the load as long as it was low in packet per second count. Sometime they just don't have a way to real world test this stuff without throwing it into the wind to see if it flies. I do have two big complaints about Broadvoice. I keep getting billed for calls that they claim are free calls on their website. I even implemented their suggested dial plan for Asterisk. Second complaint is that I can only register with one proxy at a time. Why shouldn't I be able to register with all three? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter http://www.0xdecafbad.com Sent: Monday, May 02, 2005 11:59 PM To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Anyone else having Broadvoice issues today? On Mon, 2005-05-02 at 21:39 -0700, Luki wrote: has issues for 15 minutes. Stuff happens; I guess they are implementing the required wire-tapping and/or 911 stuff... or whatever. I think its fair to say they are doing something :) I also have few problems with them in general, just stuff happened recently, and while for some things, temporary outages arent that big of a deal, for many telephone stuff is more critical to their business or whatever. VoIP in general is new, and the legislation for US based (or those that operate there) companies is changing which is going to force them to make infrastructure changes. Depending on how it was originally set up it depends on how severe that is. I am curious though about a companies competence when they have a production system and it takes a week of multiple outages to chnage something. You would think any professional company would have a test and development network seperate from the production one where they can *anounce* and schedule downtime for infrastructure changes. All in all though my only complaint with broadvoice is that their tech support knows very little on average even about broadvoice specific things, like their rate plans and what is actually included with each package (ie which exchanges are within which subpackage for a given country). To bil this information has to exist somewhere, you would think that on a corporate level they would make this information more available like vonage does, but I can live without that. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else having Broadvoice issues today?
On Tue, 2005-05-03 at 00:09 -0500, Tim Connolly wrote: No.. but... In their defense though, Cisco sold us a million dollars in routers taunting they could handle the load. 6 months later we were trading them in for Junipers because they were only able to handle the load as long as it was low in packet per second count. Sometime they just don't have a way to real world test this stuff without throwing it into the wind to see if it flies. Yes, I can see that, I dont know if that is the case or if its configuration changes. Either way they should anounce that they are upgrading if that is what they are doing. Given the short duration of the outages and their frequency over time I would expect that is what they are doing and its not failing equipment. I do have two big complaints about Broadvoice. I keep getting billed for calls that they claim are free calls on their website. I even implemented their suggested dial plan for Asterisk. Second complaint is that I can only register with one proxy at a time. Why shouldn't I be able to register with all three? dont forget about proxy.nyc.broadvoice.com which is more hidden.. proxy.lab.broadvoice.com I think its testing but who knows, broadvoice is alledgly lowell ma based (about 1 hour outside boston) and lab is near there based on traceroute. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else tried Speex 1.1 CVS?
[EMAIL PROTECTED] wrote: I built the CVS version of the Speex library - v1.2 it calls itself. Asterisk seg faults trying to use codec_speex.so. I'll have a look to try to fix it, but thought I'd just ask if anyone else knows what needs to be done? Hmm, should be OK if you're using the latest speex. There was an interim period where the API changed, such that speex_(encode|decode) was changed to take use int instead of float, but later, the int api was changed to use speex_(encode|decode)_int instead.. -SteveK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else seeing this?
Hi Brian, I haven't reported this yet, as I don't have an overall picture of what is happening, but A couple of weeks ago I had several machine lockups on the same day while testing MFC/R2 with a tor2. It hasn't happened any more here. I have no idea why it suddenly started or stopped. However, now people are starting to deploy R2, I have reports of occasional lockups with tor2 cards. I have no idea if these lockups have the same cause as mine. Regards, Steve Brian West wrote: Anything after these versions: zaptel.c version 1.95 (known working) chan_zap.c version 1.357 (known working) with a tor2 card... causes kernel panic... Can anyone else confirm this? I honestly think it's a combo issue with the new zap reload and that zaptel change. But I have spent hours trying to narrow it down to those two files and those changes. Has anyone else seen strange issues when using PRI? If we have zaptel.c 1.95 and latest chan_zap.c you can place and take calls but if you do something like show channels at the CLI you'll deadlock the box. I have no thread apply all bt since the glibc on this box didn't have debug compiled in on it. (will retry this tomorrow) If you have the latest zaptel.c and the latest chan_zap.c placing any call out/in the zap interface will cause a kernel panic and kill the box. Use the above listed known working files and you have no problems. I would open a bug report but I would like to find more information before doing so. Thanks, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else tried Speex 1.1 CVS?
On Sun, 2004-10-17 at 23:16, [EMAIL PROTECTED] wrote: I built the CVS version of the Speex library - v1.2 it calls itself. Asterisk seg faults trying to use codec_speex.so. I'll have a look to try to fix it, but thought I'd just ask if anyone else knows what needs to be done? Use version 1.0.4. Works on my box (v1-0). Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else having Broadvoice Problems?
Could this possibly indicate a problem with asterisk where the first address resolved from sip.broadvoice.com isn't functioning correctly, and asterisk doesn't try the alternate address? This isn't a scientific analysis...just proposing an ideaI don't know enough about the internals to make a statement like this, I'm hoping that someone out there can provide more intelligent analysis of this! Asterisk does no do DNS queries every time it tries to send a call to another provider. It only does so on startup/reload. So it will only pickup the first address resolved. If it is down (like in your case) then your screwed. Might be worthwile to put in a BUG report to get a better funcionality here. Like for example if the remote host does not answer then redo the DNS query... Andres ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else having Broadvoice Problems?
Could this possibly indicate a problem with asterisk where the first address resolved from sip.broadvoice.com isn't functioning correctly, and asterisk doesn't try the alternate address? This isn't a scientific analysis...just proposing an ideaI don't know enough about the internals to make a statement like this, I'm hoping that someone out there can provide more intelligent analysis of this! Asterisk does no do DNS queries every time it tries to send a call to another provider. It only does so on startup/reload. So it will only pickup the first address resolved. If it is down (like in your case) then your screwed. Might be worthwile to put in a BUG report to get a better funcionality here. Like for example if the remote host does not answer then redo the DNS query... Instead of doing another DNS query, sip.broadvoice.com is returning more than one IP address, simply keep all of the addresses returned from that initial DNS query, and if a call fails on one, just try the others already in memory. I suppose, if all of those failed, performing another DNS query to see if there are new entries to try would make scenarios where one IP address fails less likely to impact service. The system would have a little more internal redundancy, allowing it to gracefully handle failover to other addresses, making recovery from similar external failures more automated and requiring less manual intervention to restore service. Almost always a good thing for a telco system. From my experience, it isn't that unusual for multiple IP addresses to be provided for VoIP services, or for one of the IP addresses to be down from time to time for maintenance purposes. As I mentioned in a previous message, I'm just getting started with Asterisk. Perhaps something like this is already happening, though it doesn't sound like it based on what I've been reading in this thread. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else having Broadvoice Problems?
I have also been having problems today registering... I contacted them, but they have no known issues. It finally did register on it's own. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andre Normandin Sent: Wednesday, July 21, 2004 8:44 PM To: Asterisk-Users Subject: [Asterisk-Users] Anyone else having Broadvoice Problems? Suddenly my broadvoice will no longer register. It was working fine for over 1 month without a single problem, now I get a SIP registration timed out message. I called them, and I was told that they are experiencing problems, and they hoped to have it resolved ASAP. I called them at around 10 AM EST this morning. It's now 8:30 EST PM, and I still have not heard back, and the problem is not resolved. Is anyone else having problems with their broadvoice account? - Andre ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else having Broadvoice Problems?
I have also been having problems today registering... I contacted them, but they have no known issues. It finally did register on it's own. For those having trouble with BV, try this for a test: in sip.conf, replace occurrences of sip.broadvoice.com with this IP: 147.135.0.129, explanation follows... I just got off the phone with BV support who said they are having some hardware issues on the west coast which should be fixed sometime in the morning on Thursday. Interestingly, he said that despite this problem there are hundreds of asterisk users currently registering without any problems. They suggested that maybe some users are specifying IP addresses instead of FQDNs in their sip.conf, foiling BV's failover capabilities. I *do* use a FQDN and still have no outbound calling working (inbound seems to work - albeit intermittently). In my testing, I noticed that a DNS query on sip.broadvoice.com returns two addresses: 147.135.8.129 and 147.135.0.129, with the .8.129 address being returned more often than the other...if I replace references to sip.broadvoice.com in sip.conf with the second address (147.135.0.129) my outbound calling starts working again. Could this possibly indicate a problem with asterisk where the first address resolved from sip.broadvoice.com isn't functioning correctly, and asterisk doesn't try the alternate address? This isn't a scientific analysis...just proposing an ideaI don't know enough about the internals to make a statement like this, I'm hoping that someone out there can provide more intelligent analysis of this! This also makes me wonder if the hundreds of other asterisk users who are registering correctly despite the hardware problem *are* using IP addresses instead of sip.broadvoice.com and just happen to have used the IP that is still working? Marty ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else use Audacity for prompts?
Why compress all your prompts to .gsm files? Isn't * going to have to reformat them anyway based on the codec being used for the call? I have all my voice prompts as 8khz/16bit .wav files (* can't seem to play back 8 bit files). I recorded them through soundforge as a 48Khz/16bit mono .wav - did a little tweaking to compress and brighten them up - then resampled to 8khz. Quality is as good as any I've heard from any commercial PBX... It is important to me, if I'm going to use * for my business, that the voice prompts sound as clean and clear as any other system - from a marketing/PR standpoint. Dave Redmore ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else use Audacity for prompts?
- Original Message - From: Brian Capouch [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, October 06, 2003 5:21 PM Subject: [Asterisk-Users] Anyone else use Audacity for prompts? I am using Audacity to record some voice prompts. The .wav files I'm producing are of stellar quality. However, once I turn them into .gsm, they sound buzzy and muffled. I know that some of this comes with the territory, but I wonder if there is anyone out there who does this routinely, and who can advise me as to the MO I could use that results in the highest quality in the resulting playback files. Thanks. B. What are you using to convert the wav files to gsm? I've been using 'sox' under Linux and have had no quality issues whatsoever. An example line to convert: sox file.wav -r 8000 -c 1 file.gsm -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else use Audacity for prompts?
Shaun Ewing wrote: I know that some of this comes with the territory, but I wonder if there is anyone out there who does this ( wav - gsm) routinely, and who can advise me as to the MO I could use that results in the highest quality in the resulting playback files. What are you using to convert the wav files to gsm? I've been using 'sox' under Linux and have had no quality issues whatsoever. An example line to convert: sox file.wav -r 8000 -c 1 file.gsm That's the exact command I'm using, but it sure does sound crappy to my ears. Perhaps that's the best I can do w/gsm, and of course I expect once the sound is sent through the PSTN most of the highs and bottom are gone anyways. I was just hoping there was something I could do to make the resulting files a bit clearer. Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else use Audacity for prompts?
I have had been recording my gsm files by getting through to the Asterisk answering service using a GrandStream BudgeTone 102 phone. I then copy the file which is stored in voicemail and use sox to increase the volume. Results are okay but nothing to write home about particularly (or maybe that is just my lack of a good telephone voice). Michael On Mon, 6 Oct 2003, Brian Capouch wrote: Shaun Ewing wrote: I know that some of this comes with the territory, but I wonder if there is anyone out there who does this ( wav - gsm) routinely, and who can advise me as to the MO I could use that results in the highest quality in the resulting playback files. What are you using to convert the wav files to gsm? I've been using 'sox' under Linux and have had no quality issues whatsoever. An example line to convert: sox file.wav -r 8000 -c 1 file.gsm That's the exact command I'm using, but it sure does sound crappy to my ears. Perhaps that's the best I can do w/gsm, and of course I expect once the sound is sent through the PSTN most of the highs and bottom are gone anyways. I was just hoping there was something I could do to make the resulting files a bit clearer. Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else use Audacity for prompts?
On Mon, 6 Oct 2003, Brian Capouch wrote: I was just hoping there was something I could do to make the resulting files a bit clearer. put them through baudline and see what's happening also you might wanna try bandpass filter using ecasound - wasim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else use Audacity for prompts?
On 06/10/03 08:25, Shaun Ewing wrote: The .wav files I'm producing are of stellar quality. However, once I turn them into .gsm, they sound buzzy and muffled. An example line to convert: sox file.wav -r 8000 -c 1 file.gsm It'll sound much better if you go: sox file.wav -r 8000 -c 1 file.gsm resample Of course, there's only so much you can do to make 8kHz prompts sound any good. Doing the original recording at 8kHz is a good start. -- Alastair Maw System Analyst @ MX Telcom www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else use Audacity for prompts?
Why, oh why, do we have to be limited to 8kHz prompts in the first place? Alastair Maw wrote: On 06/10/03 08:25, Shaun Ewing wrote: The .wav files I'm producing are of stellar quality. However, once I turn them into .gsm, they sound buzzy and muffled. An example line to convert: sox file.wav -r 8000 -c 1 file.gsm It'll sound much better if you go: sox file.wav -r 8000 -c 1 file.gsm resample Of course, there's only so much you can do to make 8kHz prompts sound any good. Doing the original recording at 8kHz is a good start. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else use Audacity for prompts?
On Mon, 2003-10-06 at 11:08, Brad Waite wrote: Why, oh why, do we have to be limited to 8kHz prompts in the first place? Because that is what telephony is based on. 8khz by 8 bit if on a digital link and 7 bit if in RBS signaling. Why are you so worried about that amount of degradation when you can't control how much makeup and other crap is sitting in the speaker holes on the other side deadening the sound. Then there is the weather related noise introduced in old analog links, and then there is the whole analog loop. If the sound quality is of such a big issue with you, go to something other than a telephony app to use. Alastair Maw wrote: On 06/10/03 08:25, Shaun Ewing wrote: The .wav files I'm producing are of stellar quality. However, once I turn them into .gsm, they sound buzzy and muffled. An example line to convert: sox file.wav -r 8000 -c 1 file.gsm It'll sound much better if you go: sox file.wav -r 8000 -c 1 file.gsm resample Of course, there's only so much you can do to make 8kHz prompts sound any good. Doing the original recording at 8kHz is a good start. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone else use Audacity for prompts?
I have had to work on some files recently with a similar problem. It seems that when a file is recorded in 16 bit and converted to 8 bit, the clarity is lost. I have found the following ways the most productive: 1)Record through the voicemail system then import and edit them afterwards. As long as you use a good quality channel, this operates well. 2) Use sox with -t oss /dev/dsp as the input and a gsm file as the output. -r 8000 for the sample rate and -b for 8 bit. On Mon, 2003-10-06 at 17:08, Brad Waite wrote: Why, oh why, do we have to be limited to 8kHz prompts in the first place? Alastair Maw wrote: On 06/10/03 08:25, Shaun Ewing wrote: The .wav files I'm producing are of stellar quality. However, once I turn them into .gsm, they sound buzzy and muffled. An example line to convert: sox file.wav -r 8000 -c 1 file.gsm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else use Audacity for prompts?
And then use standard Unix commands to move that recording to where you want it like /var/lib/asterisk/sounds/new-recording.gsm so you can then call it from your menus or prompts. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Monday, October 06, 2003 11:28 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Anyone else use Audacity for prompts? Why do stuff the hard way? ; used to record prompts exten = 205,1,Wait(2) exten = 205,2,Record(/tmp/asterisk-recording:gsm) exten = 205,3,Wait(2) exten = 205,4,Playback(/tmp/asterisk-recording) exten = 205,5,Wait(2) exten = 205,6,Hangup On Mon, 6 Oct 2003, Stuart Mackintosh wrote: I have had to work on some files recently with a similar problem. It seems that when a file is recorded in 16 bit and converted to 8 bit, the clarity is lost. I have found the following ways the most productive: 1)Record through the voicemail system then import and edit them afterwards. As long as you use a good quality channel, this operates well. 2) Use sox with -t oss /dev/dsp as the input and a gsm file as the output. -r 8000 for the sample rate and -b for 8 bit. On Mon, 2003-10-06 at 17:08, Brad Waite wrote: Why, oh why, do we have to be limited to 8kHz prompts in the first place? Alastair Maw wrote: On 06/10/03 08:25, Shaun Ewing wrote: The .wav files I'm producing are of stellar quality. However, once I turn them into .gsm, they sound buzzy and muffled. An example line to convert: sox file.wav -r 8000 -c 1 file.gsm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
