Re: [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!

On Fri, Mar 19, 2010 at 01:26:43AM +0200, Tzafrir Cohen wrote: > On Thu, Mar 18, 2010 at 05:03:12PM -0500, Warren Selby wrote: > > On Thu, Mar 18, 2010 at 6:56 PM, Daniel Leite de Abreu > > wrote: > > > > > Hi David! > > > > > > > > > Thanks very much for helping me out will all ! > > > > > > > >

Re: [asterisk-users] Free iPhone Asterisk Function and Application Reference

On Fri, Mar 19, 2010 at 10:50:17AM -0400, Zeeshan Zakaria wrote: > Hi Matt, > This is very useful. But what about android platforms? Will it run on it? Just use an RSS reader. I guess browsers and RSS readers on the iPhone are too limited. -- Tzafrir Cohen icq#16849755

[asterisk-users] Call Drops while doing assisted transfer from remote location

Hi all, We have our system hosted publicly and 4 phones are connected remotely at employee's home, and when they try to do a assisted transfer to one of the employee at the main office, the call is lost. For ex: person A calls person B, person B calls person C for assisted transfer, and as soon as

Re: [asterisk-users] register => 2345:passw...@sip_proxy/1234

2010/3/19 tjoen : > register => tjoen:mypas...@sip_proxy/1234 > > [sip_proxy] > type=peer > host=ekiga.net I guess you need to register to the actual hostname, not the peers name. register => tjoen:mypas...@ekiga.net/1234 Chris -- ___

Re: [asterisk-users] Polycom not updating the directory list

The -directory.cfg permission is 777 with a symoblic link pointing to -directory.xml with permission of 644. I would manually edit the -directory.xml and make the changes needed. Upon rebooting the phone, the directory is still show the old contacts. Somehow the pho

Re: [asterisk-users] Live Audio Streaming- From Aux interface-Online resource

Thanks alot for the value able information. My client is not sure about the requirements as he reaches a final decision then i can move forward to start working on it. Thanks for the info. On Fri, Mar 19, 2010 at 6:30 PM, Jonathan Addleman wrote: > Philipp von Klitzing wrote: > >> I would like t

[asterisk-users] Setting Caller ID for attended transfer

Hello list, I'm sending calls to a queue in the attended way, that is, *1.* the original call is put on hold, *2.* a second line is open to call the queue, *3.*after an agent is connected the original call is transfered to its final destination. 1. Zap/1-1 <--> SIP/agentA-tag1 2.

Re: [asterisk-users] too much sockets open by asterisk

Hi: 于 2010年03月19日 22:02, Leif Madsen 写道: > Ilya Pichugin wrote: >> Hi All! >> >> I've set up Asterisk asterisk-1.6.2.2 >> >> >> My question is why asterisk opens so many sockets and does not close >> that? > > Sounds like an open bug in mantis. > https://issues.asterisk.org/view.php?id=16774 > >

[asterisk-users] register => 2345:passw...@sip_proxy/1234

sip.conf.sample: ;register => 2345:passw...@sip_proxy/1234 ; ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider ; connect to local extension 1234 in extensions.conf, default context, ; unless you configure a [sip_proxy] section below, and configure a ; context. sip.conf

Re: [asterisk-users] how to configure caller id

hi leif, thx for replying. can u plz ellabroate how to use 'o' optioan in Dial so that callerid should work. thx --- On Fri, 19/3/10, Leif Madsen wrote: From: Leif Madsen Subject: Re: [asterisk-users] how to configure caller id To: "Asterisk Users Mailing List - Non-Commercial Discussion" Da

Re: [asterisk-users] SPA3102 + asterisk drop call and loop (was SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?) )

Ok, I downgraded spa3102 to 3.3.6. Now when I make a call from pstn and call is established asterisk seems to drop the call. However I still hearing ringback on pstn side, call is established again, and asterisk drops the call again, like a loop. -- Executing [preat_ad...@nodo:1] Playback("SI

Re: [asterisk-users] Free Daily Asterisk News iPhone and iPod Touch app

Actually I might be wrong but haven't tried it yet because the download page is not available or the link is broken. I have however an iPhone too to try it. On 2010-03-19 10:16 AM, "Leif Madsen" wrote: Motorola Droid can run iPhone/iPod touch apps? Cool! :) Leif. Zeeshan Zakaria wrote: > Than

Re: [asterisk-users] Free iPhone Asterisk Function and Application Reference

Hi Matt, This is very useful. But what about android platforms? Will it run on it? On 2010-02-21 9:43 PM, "Matt Riddell" wrote: Hi all, I've uploaded a free app for the iPhone called AsteriskRef to the Apple AppStore. This allows you to lookup applications and functions using your iPhone or iP

Re: [asterisk-users] Free Daily Asterisk News iPhone and iPod Touch app

Actually I On 2010-03-19 10:16 AM, "Leif Madsen" wrote: Motorola Droid can run iPhone/iPod touch apps? Cool! :) Leif. Zeeshan Zakaria wrote: > Thanks Matt. This should be useful. I'll give it a try on my Motorola > D... >> > wrote: >> >> I've released another fr

Re: [asterisk-users] Define an array of sip number in sip.conf

Good idea Leif. On 2010-03-19 10:16 AM, "Leif Madsen" wrote: Zeeshan Zakaria wrote: > You'll have to type them all in manually. Or do what I did several > times... Using the script approach, you can generate many extensions with an #exec in sip.conf which will then trigger a script (such as via

Re: [asterisk-users] Free Daily Asterisk News iPhone and iPod Touch app

Motorola Droid can run iPhone/iPod touch apps? Cool! :) Leif. Zeeshan Zakaria wrote: > Thanks Matt. This should be useful. I'll give it a try on my Motorola > Droid/Milestone. > >> On 2010-03-18 5:31 PM, "Matt Riddell" > > wrote: >> >> I've released another free ap

Re: [asterisk-users] Define an array of sip number in sip.conf

Zeeshan Zakaria wrote: > You'll have to type them all in manually. Or do what I did several > times, write a script in php which will generate the sip.conf with that > many extensions. Even better look into using realtime architecture, > where you can quickly generate as many extensions as you l

Re: [asterisk-users] how to configure caller id

cool dude wrote: > now i want when i call from my mobile to pstn line my mobile no should > be displayed in softphone Use the 'o' option in Dial(). Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] too much sockets open by asterisk

Ilya Pichugin wrote: > Hi All! > > I've set up Asterisk asterisk-1.6.2.2 > > > My question is why asterisk opens so many sockets and does not close > that? Sounds like an open bug in mantis. https://issues.asterisk.org/view.php?id=16774 Searching for "udp sockets" in mantis produced that issue

Re: [asterisk-users] Better SIP security please! Was: (no subject)

> > Hey Philipp, > You can check out http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk for setting up from brute force detection and blocking with asterisk. There are also a link at the bottom about rate limiting registrations via iptables. -- __

Re: [asterisk-users] Live Audio Streaming- From Aux interface-Online resource

Philipp von Klitzing wrote: >> I would like to know if any one have experience with live audio >> streaming like 1. Streaming from an online resource > > Look at app_ices and icecast. > http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Ices If that doesn't work for some reason (In my case

Re: [asterisk-users] SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?)

Thanks! On Thu, Mar 18, 2010 at 5:04 PM, Joseph wrote: > On 03/18/10 16:22, Sebastian Milioto wrote: > >Somebody has 5.1.7 firmware for SPA3102? > >I'm having issues with inbound/outbound calls using asterisk through > SPA3102 > >with firmware 5.1.10. I've read it has a codec bug, since it doesn

Re: [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!

On Fri, 2010-03-19 at 01:26 +0200, Tzafrir Cohen wrote: > On Thu, Mar 18, 2010 at 05:03:12PM -0500, Warren Selby wrote: > > On Thu, Mar 18, 2010 at 6:56 PM, Daniel Leite de Abreu > > wrote: > > > > After you install the kernel source, you'll need to rerun ./configure. > > Nope. The dahdi-linux m

Re: [asterisk-users] ExtenSpy Problem [SOLVED]

Ishfaq Malik wrote: > David Backeberg wrote: > >> On Mon, Mar 15, 2010 at 1:19 PM, Ishfaq Malik wrote: >> >> >>> David Backeberg wrote: >>> >>> On Fri, Mar 12, 2010 at 11:31 AM, Ishfaq Malik wrote: You didn't mention version. Could be relevant.

[asterisk-users] R: Strange initial RING

No Gordon, the 'r' parameter isn't enabled: Dial(${TRUNK}/${EXTEN},60) Thanks, Alex -Messaggio originale- Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di Gordon Henderson Inviato: venerdì 19 marzo 2010 10:55 A: Asterisk Users Ma

Re: [asterisk-users] Strange initial RING

On Fri, 19 Mar 2010, Alexandru Oniciuc wrote: > Hello list! > >I'm having a strange problem with the VoIP Gateway that > I'm using to go on the PSTN: if the number on the other end is busy or > unavailable I hear an initial RING, generated by Asterisk from what I'm > seeing and

[asterisk-users] too much sockets open by asterisk

Hi All! I've set up Asterisk asterisk-1.6.2.2 My question is why asterisk opens so many sockets and does not close that? My diagnostics commmands: # lsof | grep asteriske | grep UDP | wc -l 1214 # ls -l /proc/`ps axuw | grep asteriskexe | grep -v grep | awk '{ print $2 }'`/fd | wc -l 1257

[asterisk-users] Strange initial RING

Hello list! I'm having a strange problem with the VoIP Gateway that I'm using to go on the PSTN: if the number on the other end is busy or unavailable I hear an initial RING, generated by Asterisk from what I'm seeing and after that the line goes down with busy signal: Here is

Re: [asterisk-users] (no subject)

On Fri, Mar 19, 2010 at 3:13 AM, Zeeshan Zakaria wrote: > Fail2ban is a must. I was a victim of such attacks, and have implemented > some other measures too, but fail2ban is a must have with the link posted by > Matt which describes how to set it up for asterisk. Make sure you put your > own ip ad