Re: [asterisk-users] srv records not being honoured properly

Brian J. Murrell wrote: > Not to this point, no. 1.4 does everything I want and if it ain't > broke, don't fix it. Well, now it's broke I guess. Still, Ubuntu still > uses 1.4 and I don't like having to maintain my own packages. Asterisk 1.6 supports proper SRV record sorting, so that the look

Re: [asterisk-users] strange h323 delay issue

2008/10/19 Mindaugas Kezys <[EMAIL PROTECTED]> > Hi, > > > > Try downgrade to Asterisk 1.4.18.1. It works for us perfectly with H323. > > > > Following versions has nasty bugs, not actually related to H323, but who > knows, maybe it will help to downgrade. > > > > Regards, > > Mindaugas Kezys > >

Re: [asterisk-users] strange h323 delay issue

Hi, Try downgrade to Asterisk 1.4.18.1. It works for us perfectly with H323. Following versions has nasty bugs, not actually related to H323, but who knows, maybe it will help to downgrade. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions Fro

Re: [asterisk-users] IP Address on CDR

Hi Sir, basically I need the public IP, of the user who registered to my asterisk and made call. users IP address might changed anytime, but need to record the IP during each call so i can be able to trace it. Regards Nhadie Alex Balashov wrote: > Define "IP address of the caller?" > > From

Re: [asterisk-users] srv records not being honoured properly

On Sun, 2008-10-19 at 13:31 +0300, Kevin P. Fleming wrote: > > Asterisk 1.6 supports proper SRV record sorting, so that the lookups > will return the correct record to the module that requested the lookup. That would be in ast_get_srv() then? In 1.4.17 I do see weight processing but not priority

Re: [asterisk-users] strange h323 delay issue

2008/10/19 Giedrius Augys <[EMAIL PROTECTED]> > > > 2008/10/19 Mindaugas Kezys <[EMAIL PROTECTED]> > >> Hi, >> >> >> >> Try downgrade to Asterisk 1.4.18.1. It works for us perfectly with H323. >> >> >> >> Following versions has nasty bugs, not actually related to H323, but who >> knows, maybe it

Re: [asterisk-users] Latency woes, qos the fix?

On Oct 19, 2008, at 1:21 AM, Alex Balashov wrote: > Stephen Reese wrote: >>> Does the latency remain more or less the same regardless of the >>> bandwidth load on the pipe? >>> >>> If so, TOS bits (what you refer to as QoS) won't help you. You've >>> either got network issues (very likely if you

Re: [asterisk-users] DAHDI and wait 'w'

Jerry Geis wrote: >-- Attempting call on DAHDI/1ww for > [EMAIL PROTECTED]:1 (Retry 1) > [Oct 16 14:36:42] WARNING[16408]: chan_dahdi.c:8132 dahdi_request: > Unknown option 'w' in '1ww' > [Oct 16 14:36:43] WARNING[16408]: chan_dahdi.c:1481 dahdi_enable_ec: > Unable to enable echo can

Re: [asterisk-users] srv records not being honoured properly

Brian J. Murrell wrote: > That would be in ast_get_srv() then? In 1.4.17 I do see weight > processing but not priority processing. I'd just go look at the code > for 1.6.1 at svn.digium.com but that server does not not seem to be > responding. Reachable but HTTP requests seem to not be completi

Re: [asterisk-users] srv records not being honoured properly

Kevin P. Fleming wrote: > I guess I'm wrong then... the weight processing is in 1.4 already, and > hasn't changed in 1.6. You are right, though, that there doesn't seem to > be any priority ordering, which seems odd because it should be simple. > I'll see about adding that to 1.6. Never mind...

Re: [asterisk-users] IP Address on CDR

Nhadie schrieb: > Hi Sir, > > basically I need the public IP, of the user who registered to my > asterisk and made call. users IP address might changed anytime, but need > to record the IP during each call so i can be able to trace it. > > Regards > Nhadie Hello, we use the following: exten

Re: [asterisk-users] Latency woes, qos the fix?

> Alex is correct. Always check thereare no half-duplex links in your > path. If you have an older dsl/cable modem or router that only has a > 10M ethernet, it is probably half. Also make certain there are no hubs > in the path. Keep in mind that colissions ar NORMAl for a hlaf duplex > connection.

Re: [asterisk-users] IP Address on CDR

Hi Sir, Tried it but it seems like it's saving the private IP. CDR(UserField)= what am i doing wrong? TIA Nhadie Stefan Schmidt wrote: > Nhadie schrieb: >> Hi Sir, >> >> basically I need the public IP, of the user who registered to my >> asterisk and made call. users IP address might change

Re: [asterisk-users] srv records not being honoured properly

On Sun, 2008-10-19 at 21:14 +0300, Kevin P. Fleming wrote: > > Never mind... I was mistaken. The srv_callback() function puts the > records returned by the DNS lookup into priority order (lowest numbers > first), Yes, I can see that function and while I have not audited it detail, it looks as tho

[asterisk-users] adding a second extension

I'm trying to add a second extension to my setup. The second device is able to successfully connect to the Asterisk server. I am unable to contact extension 101 from 102 and vise-versa. Also are my context setup logically or is there a better fashion to organize them? My error is at the bottom. He

[asterisk-users] Asterisk Problem

After installing a new box and asterisk. i have got these errors [EMAIL PROTECTED] ~]# asterisk Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory [EMAIL PROTECTED] ~]# asterisk -vr Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl

Re: [asterisk-users] adding a second extension

What is vitel-outbound?? an IP address?? And what version of Asterisk is this? Regards, Juan On Sun, Oct 19, 2008 at 3:30 PM, Stephen Reese <[EMAIL PROTECTED]> wrote: > I'm trying to add a second extension to my setup. The second device is > able to successfully connect to the Asterisk server. I

Re: [asterisk-users] Asterisk Problem

On 10/19/08, Ahmed torinto <[EMAIL PROTECTED]> wrote: > > > > After installing a new box and asterisk. i have got these errors > > > > > > [EMAIL PROTECTED] ~]# asterisk > > > > Unable to open pid file '/var/run/asterisk/asterisk.pid': > No such file or directory > mkdir /var/run/asterisk -- Kri

Re: [asterisk-users] adding a second extension

On Sun, Oct 19, 2008 at 3:55 PM, Juan Rodríguez <[EMAIL PROTECTED]> wrote: > What is vitel-outbound?? an IP address?? > And what version of Asterisk is this? > Regards, > Juan vitel-outbond is the connection to my sip provider Version 1.6 of Asterisk I'm able to make incoming and outgoing calls j

Re: [asterisk-users] Asterisk Problem

When installing Asterisk, did you issue the command "make config" after "make samples" ?? If so, try issuing "service asterisk start" on RedHat or "/etc/init.d/asterisk start" on Debian. Regards, Juan On Sun, Oct 19, 2008 at 3:50 PM, Ahmed torinto <[EMAIL PROTECTED]>wrote: > After installing a n

Re: [asterisk-users] Asterisk Problem

Thanks, but what about the both of files (asterisk.ctl and asterisk.pid) as i have done mkdir /var/run/asterisk and then [EMAIL PROTECTED] run]# [EMAIL PROTECTED] run]# asterisk -vrUnable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)[EMAIL PROTECTED] run]# >

Re: [asterisk-users] Asterisk Problem

i have done that as follow [EMAIL PROTECTED] asterisk]# service asterisk start Starting asterisk: [ OK ] [EMAIL PROTECTED] asterisk]# asterisk [EMAIL PROTECTED] asterisk]# asterisk [EMAIL PROTECTED] asterisk]# asterisk -vr Unable to connect to remote aste

Re: [asterisk-users] adding a second extension

On Sun, Oct 19, 2008 at 4:11 PM, Juan Rodríguez <[EMAIL PROTECTED]> wrote: > First, I think is better to to have SIP/vitel-outbound/${EXTEN} than > having SIP/[EMAIL PROTECTED] > And try issuing SIP SET DEBUG on the cli to see what happens when making the > call, post back what you see making calls

[asterisk-users] Got SIP response 603 "Declined" back from 81.15.xx.xx

Asterisk is behind firewall, I'm able to register with the provider. Calls are coming IN OK, but when I try to call out I got: Got SIP response 603 "Declined" back from 81.15.xx.xx -- #Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-

Re: [asterisk-users] Asterisk 1.4 and openLDAP

That looks cool. Will have a play. On 10/18/08, Ming Yong <[EMAIL PROTECTED]> wrote: > Anael, > You should take a look at Druid (Open Source Unified Communications) > Project based on Asterisk that has complete LDAP backend and Zimbra > connector. > It's an open source project & we are looking for

Re: [asterisk-users] Asterisk 1.4 and openLDAP

The LDIF needs updating as it's not a working example. I'll have one next week. I'll release an updated schema too. Gavin. On 10/18/08, Tilghman Lesher <[EMAIL PROTECTED]> wrote: > On Saturday 18 October 2008 02:30:16 Anael DIAZ wrote: >> I need help in implementing Asterisk with LDAP. I' ve inst

Re: [asterisk-users] Asterisk 1.6.1 + openais

I enabled the subscribe_event in the ais.conf and restarted aisexec. After that I restarted asterisk and the only warning I got in console was Oct 11 6:38:04.340485 [CLM ] nodeget: trying to find node If I disable the subscribe_event, asterisk starts as normal. On Tue, Oct 14, 2008 at

Re: [asterisk-users] OT: Polycom IP330 user problem

Could it be DND? I noticed the other day on my 501 that if I set do not disturb the phone still "rings" -- it is just silent. This could be caused by the configuration, I am unsure. On Sat, Oct 18, 2008 at 3:16 PM, Bill Michaelson <[EMAIL PROTECTED]> wrote: > I recently sent this email to a user i

Re: [asterisk-users] Latency woes, qos the fix?

On Sun, Oct 19, 2008 at 12:31 AM, Stephen Reese <[EMAIL PROTECTED]> wrote: > My latency is kind of high and the voice delay is noticeable. Then pretty much all you can do is lower the latency to lower the voice delay, or use a connection to th e PSTN that has a marginally lower delay if you have n

Re: [asterisk-users] Asterisk Problem

Check the permissions for the directory. Jai http://www.didforsale.com . On Sun, Oct 19, 2008 at 1:19 PM, Ahmed Torintino <[EMAIL PROTECTED]>wrote: > i have done that as follow > > > [EMAIL PROTECTED] asterisk]# service asterisk start > > Starting asterisk:

[asterisk-users] Panama DID

Looking for a low cost SIP or IAX DID in Panama, would prefer to pay based on inbound call minutes as going to be minimal inbound traffic and almost no outbound traffic. Please email pricing info. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-2

Re: [asterisk-users] adding a second extension

On Sun, Oct 19, 2008 at 5:43 PM, Juan Rodríguez <[EMAIL PROTECTED]> wrote: > Try reinstalling Asterisk, because in the channel.c this error is returned > if the channels TEC (in this case SIP) is not found. > Weird!! > Let me know if it works. > Regards, > Juan So the extensions.conf and sip.conf

Re: [asterisk-users] adding a second extension

Stephen: Your configuration files looks fine. Try from the CLI issuing "originate SIP/101 extension [EMAIL PROTECTED]", having the 101 online, then do that with "originate SIP/102 extension [EMAIL PROTECTED]". See what happens. If you got a CVS commit, commit again or try installing a release. ht

[asterisk-users] Zaptel FXO offhook when connected to PSTN

I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having an annoying issue with the FXO ports. As soon as I plug either one into the phone line it's as though the line is disconnected i.e. get disconnected tone when trying to dial out, line is busy when dialling in. The CLI shows

Re: [asterisk-users] adding a second extension

On Sun, Oct 19, 2008 at 11:21 PM, Juan Rodríguez <[EMAIL PROTECTED]> wrote: > Stephen: > Your configuration files looks fine. Try from the CLI issuing "originate > SIP/101 extension [EMAIL PROTECTED]", having the 101 online, then do that with > "originate SIP/102 extension [EMAIL PROTECTED]". See w

Re: [asterisk-users] adding a second extension

The second call its OK, so the problem it is just with the Dial(SIP/102), so try: originate SIP/102 application Dial SIP/102 and originate SIP/101 application Dial SIP/102 and originate SIP/102 application Dial SIP/101 On Sun, Oct 19, 2008 at 11:46 PM, Stephen Reese <[EMAIL PROTECTED]> wro

Re: [asterisk-users] adding a second extension

ast_request: No channel type registered for ''SIP' Notice the extra ' in the message. That is either an error in the error message or you have a an extra ' in your Dial line. Something like Dial('SIP/ I'm surprised nobody else noticed this. Stephen Reese wrote: > On Sun, Oct 19, 2008

[asterisk-users] QoS VoIP

Hi! I have some problem in my asterisk 1.4.2, I've installed it on centOS 5.2 and this didn't accept voip QoS and can't route the packets having voip QoS. So I should change voip packets to be routing with centOS. I want to use iproute2 but i don't what to do after installing iproute2. Anyone cou

[asterisk-users] asterisk setup

Hi folks, Am new to asterisk pbx systems. I am trying to figure out what to do, I'll list and folks feel free to give feedback and advice. MAIN purpose for usage: 1.exposure to setup an asterisk box 2.get home phone service via VOIP/internet connection. tasks so far --