[asterisk-users] Friday Jan 14th @ 12 Noon EST: Humbug

2011-01-13 Thread randulo
Greetings ${FellowVoIPuser}, When I saw the word "Humbug" in the Asterisk mailing list, I remembered my friend Nir Simionovich had mentioned it to me at some point, possibly at the big wine tasting party in Rostock during AMOOCON, which may explain why I had forgot about it. Seeing the thread on t

Re: [asterisk-users] Fax stopped working when upgrading to 1.8.2

2011-01-13 Thread magnus.b
Did apply the patch and did a recompile, no difference, fax still not working. But I did notice one thing, when I was standing at a fax attched to PSTN and trying to send a fax to a fax attached to the Asterisk: The PSTN fax never switched to saying “Sending...” in the display just “Dialing”, bu

[asterisk-users] 5-7 second delay in connecting outgoing FXO calls

2011-01-13 Thread ftarz
I'm running AsteriskNow 1.7.1 with a OpenVox 2FXO/2FXS card, SPA942 SIP phone and outgoing SIP and IAX routes. When I dial local PSTN numbers from the SPA942 using the FXO channels I observe a 5-7 second delay between when the PSTN number answers the call and when Asterisk connects the call at m

Re: [asterisk-users] Polycom Blf / Directed Pickup

2011-01-13 Thread Mark Murawski
Thanks! Blf is working now. I forgot I had to set set subscribecontext. When a phone is ringing, the blf light is solid red and the icon is a (/) type icon indicating unavailable. I'm also interested in directed pickup. I set up the following: call.directedCallPickupString="*6" call.dire

Re: [asterisk-users] CallerID and URL pop up for windows...

2011-01-13 Thread Bruce B
What you need already exists: http://bestof.nerdvittles.com/applications/screenpop/ But better thing would be to a have TAPI for outlook to query Outlook contact as well because it allows for making notes on the contact. I am willing to pay f

Re: [asterisk-users] Problems with ZAP Channels

2011-01-13 Thread Satish Patel
Run asterisk in verbose and and dial zap. Make sure you have hangup dialplan. -- Sent from my iPhone On Jan 12, 2011, at 1:23 PM, Antonio Modesto wrote: Hi everyone, Sometimes i am having problems with Zap channels on asterisk 1.2 (Disc-OS 1.1), after some calls, the channel continue

Re: [asterisk-users] CallerID and URL pop up for windows...

2011-01-13 Thread Danny Nicholas
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: Thursday, January 13, 2011 4:14 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] CallerID and URL pop up for windows... On Thu, 13

Re: [asterisk-users] CallerID and URL pop up for windows...

2011-01-13 Thread Gilles
On Thu, 13 Jan 2011 13:06:36 -0600, "Danny Nicholas" wrote: >Unless you need a canned app, this would be an easy program to develop on >your own. The "easiest" way (IMO) to do this would be to put a small >instance of Apache on your Asterisk server and run a CGI program that >interfaces to the lo

Re: [asterisk-users] queue_log in MySQL database

2011-01-13 Thread Tom Rymes
On 01/13/2011 2:07 PM, Tom Rymes wrote: That will require additions to your login/logout context that write entries to the log each and every time a user logs in/out. You can then report on that data. While there's a thread going on about this topic, and while I've written the above comment,

Re: [asterisk-users] queue_log in MySQL database

2011-01-13 Thread Tom Rymes
On 01/13/2011 11:25 AM, Danny Nicholas wrote: *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Thursday, January 13, 2011 10:19 AM *To:* As

Re: [asterisk-users] CallerID and URL pop up for windows...

2011-01-13 Thread Danny Nicholas
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: Thursday, January 13, 2011 11:37 AM To: Asterisk Subject: [asterisk-users] CallerID and URL pop up for windows... Anyone has a good rec

Re: [asterisk-users] queue_log in MySQL database

2011-01-13 Thread Steve Davies
On 13 January 2011 16:28, Jonas Kellens wrote: > > > I actually found this : > http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL > > But a second question : > > how can I know how long a caller stayed inside the queue untill it was > answered by a member ?? > > The queue_log table co

[asterisk-users] CallerID and URL pop up for windows...

2011-01-13 Thread Carlos Chavez
Anyone has a good recommendation for a Windows program that will open a browser URL when your phone receives a call? We had been using Yaacid but since it is no longer being developed we need to look for an alternative. It should be light weight and work on all versions of Windows. -- T

Re: [asterisk-users] queue_log in MySQL database

2011-01-13 Thread Danny Nicholas
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, January 13, 2011 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] queue_log in MySQL

Re: [asterisk-users] DTMF not being heard correctly by far end conference system

2011-01-13 Thread James Lamanna
Hi Duncan, On Wed, Jan 12, 2011 at 10:13 AM, Duncan Turnbull wrote: > Hi Thorsten > > Thanks very much, at this point my preference is rfc2833 but I will try some > other options. > > The system is generating audible tones (that I can hear), although I think > the audio is generated by the last

Re: [asterisk-users] queue_log in MySQL database

2011-01-13 Thread Jonas Kellens
On 01/13/2011 05:25 PM, James Lamanna wrote: Hi Jonas, On Thu, Jan 13, 2011 at 8:19 AM, Jonas Kellens wrote: Hello, can /var/log/messages/queue_log be saved in a MySQL database ?? So it would be easier to work with... I don't think Asterisk has this support built-in...maybe 1.8 do

Re: [asterisk-users] queue_log in MySQL database

2011-01-13 Thread Danny Nicholas
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, January 13, 2011 10:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] queue_log in MySQL database Hello,

Re: [asterisk-users] queue_log in MySQL database

2011-01-13 Thread James Lamanna
Hi Jonas, On Thu, Jan 13, 2011 at 8:19 AM, Jonas Kellens wrote: > Hello, > > can /var/log/messages/queue_log be saved in a MySQL database ?? > > So it would be easier to work with... I don't think Asterisk has this support built-in...maybe 1.8 does? However, what I do to manage queue_log is I ha

[asterisk-users] queue_log in MySQL database

2011-01-13 Thread Jonas Kellens
Hello, can /var/log/messages/queue_log be saved in a MySQL database ?? So it would be easier to work with... Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

Re: [asterisk-users] Fax stopped working when upgrading to 1.8.2

2011-01-13 Thread Vladimir Mikhelson
Magnus, Can it be the same as I experienced https://issues.asterisk.org/view.php?id=18542 ? Do not be confused by the ticket subject, it reflects the symptoms as they looked originally You can try the patch if applicable and if you know how to compile Addons in 1.8 separately or if you have

Re: [asterisk-users] OpenVPN + SIP configuration?

2011-01-13 Thread Gilles
On Thu, 13 Jan 2011 10:42:48 -0500, Bruce B wrote: >As I said, your tunnel address should be part of localnet. Otherwise you >experience what you did. Sorry about that. I didn't make long-enough calls for Asterisk to disconnect due to the lack of localnet for the VPN, and didn't know we could hav

Re: [asterisk-users] Polycom Blf / Directed Pickup

2011-01-13 Thread Sebastien Thomas
Ok, that looks good. We use FreePBX, and I know I had to modify a couple Asterisk files to get the BLF working ... here are some of my mods but may also be used for FOP2 (I dont recall which go for BLF and which go FOP2). vi /etc/asterisk/sip_registrations_custom.conf allowsubscribe=yes vi /e

Re: [asterisk-users] OpenVPN + SIP configuration?

2011-01-13 Thread Bruce B
As I said, your tunnel address should be part of localnet. Otherwise you experience what you did. -Bruce On Thu, Jan 13, 2011 at 10:00 AM, Gilles wrote: > On Thu, 13 Jan 2011 15:55:10 +0100, Gilles > wrote: > >The only issue I notice, is that Asterisk doesn't tell the other end > >when the loc

Re: [asterisk-users] Polycom Blf / Directed Pickup

2011-01-13 Thread Mark Murawski
Yeah... My directory looks like this: 62886288120010 62086208220010 62346234320010 62056205420010 62316231520010 On 01/13/2011 10:20 AM, Sebastien Thomas wrote: Is the buddy watch tag activated in your-directory.xml file ?1 Sebastien Sebastien Thomas 222 1 1 ---

Re: [asterisk-users] Polycom Blf / Directed Pickup

2011-01-13 Thread Sebastien Thomas
Is the buddy watch tag activated in your -directory.xml file ? 1 Sebastien Sebastien Thomas 222 1 1 --- Sebastien Thomas Amplisys Inc. - Digital Telephony Integration Specialists T: 514.225.4141 x222 F: 514.225.4162 TF: 1-877-AMPLISYS On 2011-01-13, at 1:32 AM, Mark Murawski wrote:

Re: [asterisk-users] OpenVPN + SIP configuration?

2011-01-13 Thread Gilles
On Thu, 13 Jan 2011 15:55:10 +0100, Gilles wrote: >The only issue I notice, is that Asterisk doesn't tell the other end >when the local end has hung up, so the other end either remains online >or hangs up after 20-30 seconds. Found it: We must add a "localnet" directive so that Asterisk hangs up

Re: [asterisk-users] OpenVPN + SIP configuration?

2011-01-13 Thread Gilles
On Thu, 13 Jan 2011 09:43:26 -0500, Bruce B wrote: >In sip_nat.conf you need to specify 10.8.0.1/24 as your localnet and also >make sure you have your externip setup as well. Else you will notice one way >audio or cut off after 30 seconds. I don't have sip_nat.conf, as I don't use any GUI to conf

Re: [asterisk-users] OpenVPN + SIP configuration?

2011-01-13 Thread Bruce B
In sip_nat.conf you need to specify 10.8.0.1/24 as your localnet and also make sure you have your externip setup as well. Else you will notice one way audio or cut off after 30 seconds. Rest of your work is all good. For security reasons the workstation that creates the keys is not connected to any

Re: [asterisk-users] OpenVPN + SIP configuration?

2011-01-13 Thread Gilles
On Tue, 11 Jan 2011 15:20:39 +0100, Gilles wrote: >By any chance, would someone have a working configuration so I can >take a look? Got it working :-) Thanks much guys for the help. For those interested, here's how I did it. Note that the appliance only has the openvpn server, so I used a Ubuntu

Re: [asterisk-users] OpenVPN + SIP configuration?

2011-01-13 Thread Gordon Henderson
On Tue, 11 Jan 2011, Gilles wrote: Hello I read a whole book on OpenVPN, but still can't figure how to configure the server + client so that the the client connects and sends SIP/RTP data through the tunnel. To get started, I'd rather use a shared key instead of X509 (certificates + keys). The

[asterisk-users] Fax stopped working when upgrading to 1.8.2

2011-01-13 Thread magnus.b
Gentlemen, We have a setup as below: PSTN – E1 – Avaya – OOH323 trunk – Asterisk – SPA-2102 – Fax machine Running Asterisk SVN-trunk-r280589M, fax working as a clock. I decided to leave “trunk” and go a stable version so I upgraded to 1.8.2. Didn’t change any config files, everything worked as b

Re: [asterisk-users] OpenVPN + SIP configuration?

2011-01-13 Thread Sebastian
On 01/11/2011 02:20 PM, Gilles wrote: Hello I read a whole book on OpenVPN, but still can't figure how to configure the server + client so that the the client connects and sends SIP/RTP data through the tunnel. To get started, I'd rather use a shared key instead of X509 (certificates + keys).

Re: [asterisk-users] Unable to get Fax t38 working with IrisTel trunk

2011-01-13 Thread Kevin P. Fleming
On 01/11/2011 04:28 PM, Karim Mardhani wrote: Hi everyone, I have been trying to get T.38 Faxing to work with Iristel sip trunks for last few days but havn't been sccussful. I am using Asterisk 1.6.2.8 and SpanDSP 0.6. Here is what I see in the tcpdump capture: You are 7 versions behind on As

[asterisk-users] WARNING T.30 ECM carrier not found

2011-01-13 Thread Flavio Miranda
CORRECTING: I have Asterisk 1.6.2.13 with TDM 410p and the Fax is connected to Dlink FXS dvg 2032s. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda From: flaviormira...@hotmail.com To: asterisk-users@lists.digium.com Date: Thu, 13 Jan 2011 09:51:24 -020

[asterisk-users] WARNING T.30 ECM carrier not found

2011-01-13 Thread Flavio Miranda
Hi list, I have search for a clear explanation about this mensage " WARNING T.30 ECM carrier not found", but until now I dont succed on it.Anybody know how can I handle with this problem? I have Asterisk 1.6.2.13 with TDM 410p and the Fax is connected to Dlink FXO dvg 2032s. Att, Flavi

Re: [asterisk-users] SetVar Warning

2011-01-13 Thread Doug Lytle
Steve Edwards wrote: I don't have a 1.4 system on hand, but 1.2 & 1.6 use set(). 1.4.x uses Set() as well. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- ___

Re: [asterisk-users] Call queues on load-balanced asterisks

2011-01-13 Thread Pan B. Christensen
Hello Mr. Liu, I tried searching for more information about FlexQueue, where to download etc. Google linked to vicidial.cn, which appears in your signature, but that page is all in chinese, and I couldn't find any english link. Where can I get more information about it? Is it a commercial prod