similar to: connect problem unless when verbose

Hi all, After running fine for a few months now asterisk seems to hang frequently , still functioning but the DAHDI channels seem busy (users report a busy signal when calling or being called) A reboot will allow it to run for another day or maybe 2 or 3 till the problem occurs again. running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel 2.6.32-5-686 i get the following

Hi, I was able to implement T122p one port PRI and was able to call out, but I am planning to use TE412p (includes echo cancellation) 4 port digital card (PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI connections) with proper hardware like dual core quadcore processor and 8gb RAM in one server? Also I was planning to implement using 64 bit architecture with Asterisk:

I have been looking for the specs (format, bit rate, ect) on custom ringtones for digium phones. Using the DPMA how would I deliver the ringtone to a digium phone? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120820/cb4927d0/attachment.htm>

Would it be possible to set the ringtone based on the number that was dialed? Example of what the goal is: Dial Denver number Incoming calls ring with ringtone 1 Dial main number Incoming calls ring with ringtone 2 We are currently using Digium D40, D50, D70 phones. -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi all, just subscribed to the list and first mail, nice to be here. Hopefully i'm in the right place for this question since i'm planning a little VOIP implementation at the moment and ran in to something while going through the shopping list. i noticed that a lot of VOIP phones have a double network interface allowing you to use only 1 LAN cable for both the phone and your

The Asterisk Development Team has announced the release of Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases have been created in response to a SIP remote crash vulnerability. Additionally, Asterisk versions 1.4.27.1, 1.6.0.19, and 1.6.1.11 also contain an SDP regression

The Asterisk Development Team has announced the release of Asterisk 1.2.37, 1.4.27.1, 1.6.0.19, and 1.6.1.11. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases have been created in response to a SIP remote crash vulnerability. Additionally, Asterisk versions 1.4.27.1, 1.6.0.19, and 1.6.1.11 also contain an SDP regression

Hi Everyone, I've got a Cisco 3810 rig with 6 analog FXS ports, and ethernet. I'm wondering in anyone has got one of these suckers to work with asterisk in such a way that each FXS port has it's own extension. It speaks SIP, and I can send calls from asterisk out to it, but can't figure out how to get it to pass username & pw to asterisk when I try to configure it as a

hi friends, i have installed asterisk,libpri,dahdi tar files in /usr/src. problem is that when i compile (./configure) asterisk-1.4.26.3, gtk+2.0.0 dependency is missing. i installed gtk from gtk.org, now when i am compiling gtk (./configure), i am getting this error message configure: WARNING: *** TIFF plug-in will not be built (TIFF library not found) *** configure: error: *** Checks for TIFF

i could add "r" option in dial command. this will generate a ringtone during connection. could i change this default ringtone? i tried indications.conf but not success. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/77a4d48f/attachment.htm>

Hi, i have a problem with iax2 and ringtone. Here is the call path pstn -> asterisk -> iax -> firefly or any iax phone. My problem is when i receive a call on my iax phone, the ring sound is very distort and bad. If i open my sip phone, and receive a call from my pstn, the ring is like dring dring, very normal. Otherwise, it is like a machine gun with iax Help would be really

Hi All I have been looking around and haven not been able to find a working example I have a fresh/new install of Asterisk 1.6.2.0 with dahdi 2.2.1 and libpri 1.4.10.2 I use a sangoma A200 card so I am using wanpipe 3.4.7 If I use zaptel which I read I need for app_rxfax then asterisk crashes with segfaults on startup asterisk[2624]: segfault at 30353466 ip b7eb538b sp bffda26c error 4 in

Hi, Snom phones firmware 5.0 is now out. Try it if you like: http://www.snom.com/wiki/index.php/Main_Page. Regards, --------------------------------------------------------------------- Usman Tahir snom technology AG www.snom.com --------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have uploaded a custom ringtone to our SPA-922's for the purpose of sounding like a door bell chime when the doorbell is pressed. I am using __alert_info to set this ringtone. It appears that I can only set the ringtone via alert_info for the ringtones that come from Linksys. Has anyone else seen this issue? I am doing the following: exten => 100,1,SetVar(_ALERT_INFO=doorbell)

We can use the Dial() command with the 'r' option in order to generate the UK ringtone (as we are UK based the default is UK). How do we generate a USA ringtone for example? I have tried setting the CHANNEL(language) and CHANNEL(tonezone) to 'us' (and calling Progress() beforehand) and although this works for Playtones() the Dial command still continues to play the UK ringtone.

Hi, I'm having trouble setting the ringtone on my Polycom 501. The relevant entry in extensions.conf is: exten => 801,hint,SIP/creative1 exten => 801,1,SetVar(ALERT_INFO="Test") exten => 801,2,Dial(SIP/creative1,20,Ttr) In the sip.cfg: <alertInfo voIpProt.SIP.alertInfo.1.value="Test" voIpProt.SIP.alertInfo.1.class="13"/> and <TEST

The document describes search(query, options) sort: An array of SortFields describing how to sort the results. I have created index with two fields: ''file'' and ''content'' When I give SortField name as ''file'' while searching, it results into error. The exact command given by me: index.search_each("sleepless AND dreams", :num_docs

https://bugzilla.samba.org/show_bug.cgi?id=2847 ------- Comment #2 from seregino11@gmail.com 2007-04-28 12:30 MST ------- Created an attachment (id=2558) --> (https://bugzilla.samba.org/attachment.cgi?id=2558&action=view) Mp3 Ringtone Mp3 Ringtone -- Configure bugmail: https://bugzilla.samba.org/userprefs.cgi?tab=email ------- You are receiving this mail because: ------- You are

Hello list, I'm using the following macro when calling an external callphone/GSM number : [macro-press1] exten => s,1,NoOp() exten => s,n,Playback(/var/lib/asterisk/sounds/prompts/press1) exten => s,n,Read(INPUT,,1,1,1) exten => s,n,NoOp(input : ${INPUT}) exten => s,n,GoToIf($["${INPUT}"=="1"]?exit:hangup) exten => s,n(exit),NoOp(call accepted) exten

Hi all, I have TDM2406E with 24FXO ports connecting to 10 POTS line sitting in my office. the out going calls symptom like when called party pickup the phone but the calling party still hearing the ring tone from the IP phone. Please light me up. it been many sleepless night by googling around trying to get the right answers. The digium card running on Intel 915G chipset. Below are my zaptel