Re: [asterisk-users] Question regarding keywords in sip.conf/users.conf

Unfortunately RealTime isn't going to be an option - it's another level of configuration I want to avoid, but more importantly since I'm planning on being able to run these scripts on an Astlinux install, there won't always be a MySQL database available. If worst comes to worst, and the extra

Re: [asterisk-users] SPA3102 interdigit timers bug?

2008/11/1 Rodolfo Alcazar Portillo <[EMAIL PROTECTED]>: > Hi. I have a SPA3102 updated with with Software Version: 5.1.7(GW). > > I have this settings on Voice/Regional: > > Interdigit Long Timer: 10 > Interdigit Short Timer: 3 > > Anyway, when hooking up (without dialing anything), the timeout st

Re: [asterisk-users] Hang up detection with TDM400P and Telewest/Virgin Media line

robb wrote: > I have a TDM400 working quite well, Digium dialled in and recompiled > chan_zap with some changes , to get BT Callerid working and I have > set hangup on polarity in the zaptel.conf which seems to work well > > this is a BT home line, not business, if you have a business line you >

[asterisk-users] loading misdn.conf strange error regarding out of range

Hello all! I just noticed, that since installing the latest SVN branch (152803), I receive the following error, when loading/reloading the misdn.conf file misdn reload [...] [Nov 3 16:20:37] WARNING[5267]: misdn_config.c:938 _build_general_config: misdn.conf: "misdn_init=/etc/misdn-init.conf"

Re: [asterisk-users] say load new

Hello! On Mon, Nov 3, 2008 at 6:30 PM, [EMAIL PROTECTED] <[EMAIL PROTECTED]>wrote: > I would like to use say.conf settings but every time i restart > asterisk i have to load manualy "say load new" is there a way to do it > automaticaly i use asterisk 1.4.19 > There is option in say.conf to do it

Re: [asterisk-users] Call problems

Emmanuel Pascal Bruno wrote: > I have tried that too with no results > > > On Sun, Nov 2, 2008 at 1:30 PM, Rob Hillis <[EMAIL PROTECTED] > > wrote: > > Emmanuel Pascal Bruno wrote: > > I have turned off firewall on the linux box, I have turned off > > fir

Re: [asterisk-users] help with debugging phone call

Jerry Geis wrote: > I am running 1.4.22. > > I am doing a simple call into the dialplan and am getting a strange > error: > > [Nov 3 08:32:27] NOTICE[8022]: chan_sip.c:14316 > handle_request_invite: Failed to authenticate user "404" > ;tag=547521CB-DB0D6130 > > This is the only line that prints

[asterisk-users] Call quality issue across VPN-> POTS vs SIP

Hi All, Got a strange (at least IMHO) issue that doesn't make much sense to me. Basic configuration is two sites with a site-to-site (aka router-to-router) VPN. Both sites have Cisco 7961G phones [with SIP firmware] on users' desks, and the only VoIP is internal - all of our outward telecom is

[asterisk-users] help with debugging phone call

I am running 1.4.22. I am doing a simple call into the dialplan and am getting a strange error: [Nov 3 08:32:27] NOTICE[8022]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user "404" ;tag=547521CB-DB0D6130 This is the only line that prints on the console... Typically I get

[asterisk-users] say load new

Hello all, I would like to use say.conf settings but every time i restart asterisk i have to load manualy "say load new" is there a way to do it automaticaly i use asterisk 1.4.19 Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital

[asterisk-users] Looking for a web video phone?

Is there anything like that? Any experiences? Sincerely, Robert Augustyn www.linqone.com <>___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update opti

Re: [asterisk-users] Call terminates after 20 minutes

Go to sip.conf. Find the SIP Session-Timers section. Ensure that you have this option set: session-timers=refuse This might help. If not, try other variations of the session-timers value. The default session-timer is 10 minutes - exactly half of what you claim is your duration maximums,

Re: [asterisk-users] Call quality issue across VPN-> POTS vs SIP

On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote: > Any ideas why the audio quality would be so markedly different when > the only thing that seems to be different is where the call is > originating from (POTS line vs. SIP phone)? Is it possible that calls from your POTS line are going

Re: [asterisk-users] giving a user asterisk CLI access: how bad could it get

On Nov 1, 2008, at 5:15 PM, Tilghman Lesher wrote: > On Saturday 01 November 2008 18:52:41 Alexander Lopez wrote: >> No need to compile "!" out of asterisk source >> >> Just put SHELL=/bin/false in your login script >> >> The ! command will not work... > > That's not completely true. The

Re: [asterisk-users] Looking for a web video phone?

On Nov 3, 2008, at 12:11 PM, Robert Augustyn wrote: Is there anything like that? Any experiences? X-Lite is a free download and has video capabilities. Fred Posner [EMAIL PROTECTED] Main: +1 (212) 937-7844 Direct: +1 (503) 914-0999 www.teamforrest.com smime.p7s Description:

Re: [asterisk-users] Looking for a web video phone?

Thank you, How do I embed it into the web site though? robert _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fred Posner Sent: Monday, November 03, 2008 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Looking for a web

Re: [asterisk-users] CLI dial and echo recorder

Say any ideas how to do the following from the cli In order to test I would like to dial my phone from the Asterisk cli and then record my voice on asterisk and have it played back to me? Also how can a I specify a specific callerid? Thanks Shaun ___

Re: [asterisk-users] Call quality issue across VPN-> POTS vs SIP

Bob, It's conceivable, but how would I verify this and how would I change it if that was the problem? The site that those calls terminate at is using an Asterisk Appliance so most of the config is "done for you" but it is possible to tweak the underlying configuration files (and I also have SS

Re: [asterisk-users] Blank Voicemail.Conf after Password Change

Rob, Thanks for your time and assistance. The directory is owned by asterisk and permissions seem fine there too: drwxrwxr-x 9 asterisk asterisk 3072 2008-11-03 13:12 . I never see a voicemail.conf.new created. I have done a locate on the whole server and don't see a blank one there either whic

[asterisk-users] busylevel question

I am running asterisk 1.6.0.1. I have a Grandstream GXP280 phone I use for testing. In addition I register a zoiper SIP soft phone. For the Grandstream I have busylevel=1 in sip.conf. If I place a call from the GXP280 to zoiper and then put that call on hold from the zoiper side and then call GXP

Re: [asterisk-users] CDR Posting Delay

On Oct 31, 2008, at 11:13 AM, Douglas Garstang wrote: > We have a situation where it's sometimes taking Asterisk 17-19 > minutes to post CDR's, both over the AMI, and over the MySQL socket. > It seems however that they are logged locally to /var/log/asterisk/ > cdr-csv/Master.csv right after

Re: [asterisk-users] Ztdummy and Asterisk

On Sat, Nov 01, 2008 at 11:03:09PM -0200, Aldo D. Sudak wrote: > Loading zaptel hardware modules: ztdummy. > Running ztcfg: done. There is no need to run ztcfg if you just want to use ztdummy. The call to ztcfg is probably just part of some standard init script. -- Stefan Tichy ( asterisk2 at

Re: [asterisk-users] Call quality issue across VPN-> POTS vs SIP

Just as an interesting follow-up/additional information, if I place a call to Site 2 on a POTS line, someone at Site 2 answers the call (using one of the Cisco phones) and then transfers it to me across the VPN the call sounds fine. So I think Bob's question was on the right track with it being

Re: [asterisk-users] Call quality issue across VPN-> POTS vs SIP

On Mon, 2008-11-03 at 13:17 -0500, Lincoln King-Cliby wrote: > It's conceivable, but how would I verify this and how would I change > it if that was the problem? There's a few things you can do here. 1) Check the sip.conf on both sides to see what is defined there for the trunk. Look for some dis

Re: [asterisk-users] VoIP traffic shaping

Kristian Kielhofner wrote: > On 11/1/08, OCG Technical Support <[EMAIL PROTECTED]> wrote: > >> >> This was so interesting I had to move it to its own thread! >> >> >> Is anyone using this script? How does it perform compared to the older >> WonderShaper script? >> >> > It was based off Won

[asterisk-users] Ztdummy and Asterisk

Hi again, Thanks for your answer, Stefan. In fact, ztcfg is automatically running from the startup script. Anyhow this seems harmless and not the cause of the problem. I have confirmed this with two other Asterisk servers, both having a zaptel card installed in them, but one running the same

[asterisk-users] asterisk src=dst

Hi all I saw in the CDR stocked in mysql as well as those in the csv file that some time, the src field is the same as the dst field which is the extension. When does it happens. Here, we have 4 dgits extensions and most of the time the dst field is the extension and the src field is the 10 dig

Re: [asterisk-users] Blank Voicemail.Conf after Password Change

On Monday 03 November 2008 12:19:31 Leah Newmark wrote: > The directory is owned by asterisk and permissions seem fine there too: > drwxrwxr-x 9 asterisk asterisk 3072 2008-11-03 13:12 . > > I never see a voicemail.conf.new created. I have done a locate on the > whole server and don't see a blank o

Re: [asterisk-users] Question regarding keywords in sip.conf/users.conf

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rob Hillis wrote: > Unfortunately RealTime isn't going to be an option - it's another level > of configuration I want to avoid, but more importantly since I'm > planning on being able to run these scripts on an Astlinux install, > there won't always

Re: [asterisk-users] Question regarding keywords in sip.conf/users.conf

Barry L. Kline wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Rob Hillis wrote: > >> Unfortunately RealTime isn't going to be an option - it's another level >> of configuration I want to avoid, but more importantly since I'm >> planning on being able to run these scripts on an As

Re: [asterisk-users] Question regarding keywords in sip.conf/users.conf

Historically Asterisk's config file parser ignored unknown keywords. This is useful for exactly the things you are trying to do. I hope 1.6 did not remove this "feature". Rob Hillis wrote: > Barry L. Kline wrote: >> -BEGIN PGP SIGNED MESSAGE- >> Hash: SHA1 >> >> Rob Hillis wrote: >>