[asterisk-users] wctdm24xx IRQ missing

Hello all, I just instaled a tdm2400 Digium card on my asterisk box. When it boots, I can see some error messages in dmesg. wctdm24xxp :21:08.0: Missed interrupt. Increasing latency to 8 ms in order to compensate. wctdm24xxp :21:08.0: Missed interrupt. Increasing latency to 10 ms in

[asterisk-users] Asterisk Log viewer

Hello, I want to analyze the asterisk logs files, looking for all kind of errors, ¿Anyboby knows any asterisk logs analyzer? Thanks all, Voipcrazy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

[asterisk-users] Switchboad like application

Hello all, Anybody could point me any clue about an Open Source or licensed switchboard for my users? ARI or FOP is not enought for my users. Thanks in advance. VoipCrazy -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] RTP ports

Hello, I need to limit the RTP ports used by an asterisk in a client, Actualy the range defined is from 1 to 2 udp ports. If I only have 10 local sip extension ¿how many ports/range should I set up in /etc/asterisk/rtp.conf? Which is the way to calculate the rtp ports needed on an

[asterisk-users] Snom Provisioning

Hello all, I've to deploy about 200 snom320 phones on a instalation. Do you know any knid of tool to help me with this amount of phones? I'm thinking in a provisioning tool which I use for setting up the phones. Any clue would be welcomed. Thanks. Voip-Crazy

[asterisk-users] Billing applications

Hello all, I want to instal a Billing solution in the same asterisk's box. I have browse for ast2bill asterisk billing, astercc, and more, bu ti do not know which will be the best for me. The only things i need, are, - Postpaid and prepaid applications. True CDR,

[asterisk-users] Billing applications

Hello all, I want to instal a Billing solution in the same asterisk's box. I have browse for ast2bill asterisk billing, astercc, and more, bu ti do not know which will be the best for me. The only things i need, are, - Postpaid and prepaid applications. - True CDR. Better that

Re: [asterisk-users] onnecting two asterisk using B410p BRI cards

I just plug the junper in NT mode with no success. VoipCrazy 2009/8/15 Paul Hales pdha...@optusnet.com.au: Use a standard network cable - but you have to activate the 'terminate' jumper on the NT end. - Also, the new BRI stuff in dahdi is much easier to work with than misdn. PaulH voip

[asterisk-users] onnecting two asterisk using B410p BRI cards

Hello all, I'm trying to conect two asterisk servers using two B410p Digium cards. One card on each server. I just setting up the first BRI port on server A as nt_ptp and the first BRI port on server B as te_ptp. I use an ethernet wire to connect the first port of server A (nt_ptp) with the first

[asterisk-users] Printing faxes

Hello list, I have an asterisk / hylafax / iaxmodem configured in one machine. All is working nicely. Now I need the fax to be print when arriving. ¿Anybody have this feature implementing in their systems? ¿How is the best way to get that? Any clue will be welcomed. Thanks. VoipCrazy

[asterisk-users] Webcall app needed

Hello all, I need to configure an application which let me to call from a web page. Someone has experience using apps to make webcalls? Which software do you use? Thanks. VoipCrazy. ___ -- Bandwidth and Colocation Provided by

[asterisk-users] Hylafax asterisk iaxmodem problem

Hello all, I have an asterisk box running in a customer with Hylafax, iaxmodem, asterisk 1.2.18. The service can receive faxes, from a lot of fax machines, but there are a couple of them that asterisk Hylafax cannot complete. This calls arrive the asterisk box, asterisk detect that this calls

[asterisk-users] WebCall application

Hello list, Does anybody know any free WebCall solution to let our customer call us directly via our web site? Any clue will be welcomed. Thanks. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users

[asterisk-users] B410p question

Hello list, I have got an asterisk box installed working ok with an b410p card to make and receive isdn calls. All works ok, but when a call is answer and the person starts to speak, always I can ear a beep during the call. This beep is ear some times in about 30 seconds between each beep.

[asterisk-users] Asterisk Queue question

When the asterisk a queue reset their counters? I 'm talking about this kind of info in asterisk console. show queue 600 600 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime), W:0, C:14, A:8, SL:0.0% within 0s I just say that because I have a queue with strategy Fewest

[asterisk-users] Sip Header Help

Dear List: I need to make a sip phone (spa942) answer a call but the phone must no ring. The user only has to show the callerId on the phone screen without any sound. How could I make that in asterisk? I tried to use Sip headers but I do not know how must I say the phone don't ring when

Re: [asterisk-users] Gateway errors

on, do this in the first of sip.conf file Best Regards On Mon, Sep 1, 2008 at 11:32 AM, voip crazy [EMAIL PROTECTED] wrote: Hatem, I cannot understan exactly what you told me. Could you try to explain that in other words. Better if you could post an example of this SIP trunk. thanks

[asterisk-users] Gateway errors

Hello list, I have an asterisk instalation with a bad internet connection cause this connection is down sometimes. When the connection is down and asterisk cannot get internet connection. All the extensions log out from the asterisk machine, and nobody can make any call. ¿Why if internet

Re: [asterisk-users] Gateway errors

issues with your router? Can you reach all the boxes in your lan while you are experiencing this downtime? voip crazy wrote: When I say extensions, I say extensions in the lan not in wan Thanks. VoipCrazy. 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Hello, By people do you mean people

Re: [asterisk-users] Gateway errors

When I say extensions, I say extensions in the lan not in wan Thanks. VoipCrazy. 2008/9/1 Igor Hernandez [EMAIL PROTECTED]: Hello, By people do you mean people in the lan or external users? Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com voip crazy wrote

Re: [asterisk-users] Gateway errors

Hatem, I cannot understan exactly what you told me. Could you try to explain that in other words. Better if you could post an example of this SIP trunk. thanks in advance. Voip Crazy 2008/9/1 hatem moiz [EMAIL PROTECTED]: Asterisk is looking for a SIP trunk if you have recorded the usage

[asterisk-users] Outgoing calls

Hello list, How could I limit the outgoing calls for one trunks easily? Thanks VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now:

[asterisk-users] Cisco vs Asterisk

Hello all, A client of us, is thinking to migrate their actual PBX to a Cisco CallManager. We want to sell him an asterisk box to complement the Cisco PBX. I think to use asterisk as a Voicemail server (Replazing the Cisco Unity) Has asterisk all the functionalities to replace a CIsco Unity

[asterisk-users] Asterisk dimensioning

Hello all, I need to install asterisk for 900 sip users with 2 PRI ports. It is posible to handle this number of calls/extensions with only one asterisk machine? Which is the best way to install that? two asterisk with openser. One asterisk with openser . Is it necesary run a SER server on

Re: [asterisk-users] Asterisk dimensioning

guys to describe the ideal setup you should use. I have an idea of how I might do it, but I wouldn't want to get it wrong. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of voip crazy Sent: Wednesday, July 09, 2008 3:01 AM To: Asterisk Users

[asterisk-users] Removing voicemail messages

Hello, I want to create an script which remove all the old voicemail messages. I make a simple Bash script to delete all the new messages for the extension 100. Something like, rm /var/spool/asterisk/voicemail/defaul/100/INBOX Should I update any index file or something after reemove them?

[asterisk-users] Manager proxy

Hello all, Some one is using asterisk and queuemetrics connected via astmanproxy? How about your experience? Which proxy do you use in this kind of connection? In my instalation asterisk and Queuemetrics are installed on diferent machines and I want to avoid manager problems Thanks in advance.

[asterisk-users] Softphone accepting sip messages

Hello all, Someone knows any softphone which accept messages using sipsak? I just tried X-Lite and portsip without success Thanks Voipcrazy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25

[asterisk-users] Transfers with TE12xp

Hello all, I have an asterisk PBX working perfectly, and the transfers between extensions, works ok. The problem, when I receive a call from the line connected to the TE12Xp, and I try to transfer it, the calls hangs up. I have other analog lines and I can tranfer all the without problems. I've

Re: [asterisk-users] Transfers with TE12xp

More info about the problem. This occurs, when I try to transfer using the *2 funcionality into aterisk Thanks 2008/6/16 voip crazy [EMAIL PROTECTED]: Hello all, I have an asterisk PBX working perfectly, and the transfers between extensions, works ok. The problem, when I receive a call

[asterisk-users] Dial command and its g option

I need to execute an action after a call is hangup. I just see the command Dial has an option for that, the g option. I configure the dial command as exten = s,n,Dial(SIP/100,100,Ttg) How should I add the line which the command will be executed after the dial command in this example? I don`t

[asterisk-users] AGI after Hangup

Which is the way to run an AGI after hangup a call? The problem I have is when the call dies the AGI dies too I try the Dial command g option, but it does not work for me Any clue will be welcomed. Thanks VoipCrazy ___ -- Bandwidth and Colocation

Re: [asterisk-users] AGI after Hangup

Thanks for your answers, DeadAGI was the solution. Thanks again. Voipcrazy 2008/6/12 Andrea Cristofanini [EMAIL PROTECTED]: You have to run DeadAGI, in h . Regards Andrea Cristofanini voip crazy ha scritto: Which is the way to run an AGI after hangup a call? The problem I have is when

[asterisk-users] PRI error cause hangup calls

Dear all, When I make a call using my PRI line, all goes well, but suddently the call hangs up. I searched the asterisk logs, and I found that. Write to 55 failed: Unknown error 500 Short write: 0/15 (Unknown error 500) What does this mean? Why this occurs? How could I solve that? Someone

[asterisk-users] Sending SMS

Hello all, I want to send SMS using asterisk, I just read there are lot of apps to do that, but I do not know which to choose, like cmd SMS, Fast SMS, ZIM-SMS,.etc. http://www.voip-info.org/wiki/view/SMS Which is the way you use to send SMS messages using asterisk? Which apps do you use?

[asterisk-users] Weird Zaptel sound after anwser calls

Dear list, We have an weird problem with our FXO card (TDM01B). When I made a call using this channel, all goes well, the called phone rings but when the called phone answers the call. In me handset I can hear an weird sound like a Clack. I tryed diferents TDM cards and modules, and my

Re: [asterisk-users] Weird Zaptel sound after anwser calls

I forgot to say that I'm using bristuff-0.4.0 with zaptel 1.4.4, libpri 1.4.1 and asterisk 1.4.9 Thanks. 2008/2/22, voip crazy [EMAIL PROTECTED]: Dear list, We have an weird problem with our FXO card (TDM01B). When I made a call using this channel, all goes well, the called phone rings

[asterisk-users] Hardware needed

Dear List, I have to plan an instalation of an asterisk box for over 400 extensions (Sip and Iax2) and 4 PRI channels. I do not know which hardware (server) should I buy to support this amount of extensions. Someone made a similar instalation? which hardware (server) did you use? Which was the

[asterisk-users] Attendant phone

Dear list, I need to buy a phone which could monitor the state of the maximun number of sip extensions about 200. It is for an attendant. I just saw Snom 370 with keypad and Linksys 962 but they do not let me to monitor 200 extensions states adding keypads. Do you know any kind of phone that let

[asterisk-users] Asterisk and fax

Dear list, I need to setup asterisk to send and receibe fax. I just looking about SpanDSP, Hylafax/Iaxmodem, AsterFax,...etc. The asterisk box has Digium hardware, one TE420B and one TDM2402 (8 FXO ports). I just read the SpanDSP (txfax and rxfax) makes the system more unstable that

Re: [asterisk-users] Asterisk and fax

I want to receibe the fax via mail and send faxes via web interface and a digital send and receibe fax list. Voipcrazy 2008/2/13, Giorgio Incantalupo [EMAIL PROTECTED]: Hi VoIPCrazy, why don't you use an ATA device such as Grandstream 486 or similar? Giorgio Incantalupo voip crazy wrote

[asterisk-users] PRI with 20 channels

Dear all, I have got a PRI line with E1 20 channels, my question is: span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 If this is my zaptel config for an E1 PRI line, Which would be the config for a reduced PRI line for 15 channels? and for 20 channels? Thanks in advance. VoIPcrazy

[asterisk-users] Qsig link

Hello all, I need to conect an Asterisk with an Alcatel OmniPBX 4400 using an E1 port. It is the first time I make this kind of connection and I do not know exactly how to get it working. Someone has experience with this kind of connection? Could you paste a zapata.con and zaptel.conf files with

Re: [asterisk-users] Qsig link

) voip crazy wrote: Hello all, I need to conect an Asterisk with an Alcatel OmniPBX 4400 using an E1 port. It is the first time I make this kind of connection and I do not know exactly how to get it working. Someone has experience with this kind of connection? Could you paste a zapata.con

[asterisk-users] Snom 370 buton Recordings

Hello all, I am using the Snom 370 phone with firmware Snom370-SIP 7.0.17* *connected to an asterisk 1.2.14 and I can't record any calls using the Recording button on this phone. The extension I configured on this phone has the values Recording on demand, an the voicemail enabled. I am using

[asterisk-users] Change the Voice promps in asterisk 1.4

Hello all, Which is the best way to change the default Voice promps in asteriosk 1.4from english to french? And if I would like to add a new Voice promp set, how is the way to do? Thanks in advance. VoipCrazy ___ --Bandwidth and Colocation Provided by

[asterisk-users] Snom 320 with TDM02B and echo problems

Hello all, I instaled an asterisk 1.2.X, with two lines FXO and two snom 3200 phones, on an amd_64 processor. All goes well, the voice is clear on the remote side but in the Voip side, where the Snom 320 is placed, I hear my voice, but don't in the line, the echo is on the phone. I just play with

[asterisk-users] How to delete voice mail messages?

Hello all, Could I create a script to delete the first messages on my voice mail? In this script should I update any messages index file or there isn't any file to index them? Could you share any script to do that? Thanks in advance. VoipCrazy. ___

Re: [asterisk-users] asterisk hylafax iaxmodem

Witch interface are you using to send faxes, SIP, IAX, ZAP, MISDN,...,etc VoipCrazy 2007/10/17, Giedrius Augys [EMAIL PROTECTED]: CountryCode:1 AreaCode: 800 FAXNumber: +3705203230 LongDistancePrefix: 1 InternationalPrefix:011

Re: [asterisk-users] Eicon Diva and incoming Fax

Dear Armin, This solve my problem, when I set softdtmf and relaxdtmf to off, my asterisk machine starts to detect the incoming fax calls. Thank for your help. VoipCrazy. 2007/10/15, Armin Schindler [EMAIL PROTECTED]: On Mon, 15 Oct 2007, voip crazy wrote: Dear Armin, Bellow I send you

[asterisk-users] Eicon Diva and incoming Fax

Hello all, I am trying to set up asterisk and hylafax to send and receibe fax. The machine is connected to the PSTN is an Eicon Diva BRI (1 BRI port). My problem is that , when I send a Fax from the PSTN to this machine, the asterisk or diva or hylafax, does not detect this call as a fax and

Re: [asterisk-users] Eicon Diva and incoming Fax

[EMAIL PROTECTED]: Hello VoipCrazy !? On Mon, 15 Oct 2007, voip crazy wrote: Hello all, I am trying to set up asterisk and hylafax to send and receibe fax. The machine is connected to the PSTN is an Eicon Diva BRI (1 BRI port). My problem is that , when I send a Fax from the PSTN

Re: [asterisk-users] Eicon Diva and incoming Fax

* setting available. ;mohinterpret=default ;Asterisk 1.4: default music on hold class when placed on hold. ;qsig=on ;enable use of Q.SIG extensions. --EOF-- 2007/10/15, Armin Schindler [EMAIL PROTECTED]: On Mon, 15 Oct 2007, voip crazy wrote: Dear

[asterisk-users] Virtual server Solution

Hello all, I'm looking for a solution to offer Virtual PBX, to my clients. I just saw software with multi-tenant support and I tested it, but no one likes me enought. Finally, I want to offer this service like a kind of hosting. Has you experience with multi-tenant software? Which has you

Re: [asterisk-users] Virtual server Solution

Dear Tzafrir, I just try Destar, but one thing I dislike was, that there are no posibilities to login the manager of each virtual PBX. Then customers cannot manage their owns PBX. VoiPCrazy 2007/9/24, Tzafrir Cohen [EMAIL PROTECTED]: On Mon, Sep 24, 2007 at 11:38:38AM +0200, voip crazy wrote

[asterisk-users] Hfcmulti and B410P Digium Card

Hello all, I am getting the following error in /var/log/syslog. I have got 2 B410P cards in this box. Sep 19 17:13:31 localhost kernel: hfcmulti_rx: fifo(0) reading 128 bytes (z1=0153, z2=00d3) TRANS Sep 19 17:13:31 localhost kernel: hfcmulti_tx: fifo(0) has 382 bytes space left (z1=0013,

Re: [asterisk-users] Hfcmulti and B410P Digium Card

Maybe I find the problem, It could be cause debug is enabled. Tomorrow I will change debug to disable and I will tell you the results. Regards. VoipCrazy 2007/9/19, voip crazy [EMAIL PROTECTED]: Hello all, I am getting the following error in /var/log/syslog. I have got 2 B410P cards

[asterisk-users] Email to Voice

Hello all, Anyone knows any solution (Comercial or Free) to listen my email via a phone call? Thanks in advance. VoipCrazy ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update

[asterisk-users] Queues monitoring software

Hello all, A client of us, needs a queue monitoring system. In realtime he needs to now the PRI status, the agents logged in and logged out, the number of received calls by agent, ,etc. I am not a call center specialist and i want to find a call center software to offer to my client that

[asterisk-users] Hi ability solution

Hi all, On one of our client, I must to install an asterisk over a hi ability cluster. I have no experience with clusters an linux neither asterisk. Someone has installed an asterisk in a hi-ability enbviroment? How do you install the cluster? Witch solution did you use? Witch is the best

Re: [asterisk-users] Hi ability solution

I would say High Availability, sorry for my english. Any High availiability solution for asterisk? VoipCrazy 2007/6/25, Steve Totaro [EMAIL PROTECTED]: voip crazy wrote: Hi all, On one of our client, I must to install an asterisk over a hi ability cluster. I have no experience

[asterisk-users] Astmanproxy

Hello all, Some of you are using astmanproxy with asttapi or activa TSP? How does you make to work? Thanks VoipCrazy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] asterisk TAPI interface

Hello, I need to connect asterisk 1.2.16, with a Contect Center software that works with TAPI. As I know, asterisk doesn't support TAPI directly, if needs a tirth party software. I just reading about asttapi and Activa TAPI. does anyone test this software? have you using asterisk againts a TAPI

[asterisk-users] Internet gateway problem

Hello all, I have got an asterisk server in my LAN, getting access to internet trought a router. I have observed in my asterisk box, when the internet connection in down, the phones can not register to my asterisk. It is like chan_sip, does not work without an internet connection. If when the

[asterisk-users] Nagios asterisk monitoring

Dear list, I am trying to configure the nagios plugin called check_sip. I just read the README file included with the plugin. I follow the readme steps to configure the plugin, without success. In the nagios web interface I can see (No output!) In the status information column. If I run the

[asterisk-users] Problem with a Zap channel

Hello, I have got two zap channels configured in our asterisk server, one of them is connected to the PSTN directly and the other one is connected to a gsm track, only for mobile calls. Both of them are basic lines. I just connect an iax softphone (idefisk) to the asterisk PBX. If I make a

[asterisk-users] Asterisk callerID

Hello all, Recently I just instaled asterisk-1.2.14, zaptel-1.2.12, libpri-1.2.4 and Freepbx v.2.2.0. My zapata.conf look like this, (Pasted bellow) The problem is that the asterisk never send the callerID to the phones. I just take a look to the cdr database an there is no callerid too. I do

[asterisk-users] Recomended POE Phones

Hi all, I am looking for phones witch support POE, with a good relation between quality and price to work with asterisk. I just see the Thompson st2030 and the Linksys SPA 922 an SPA 942. Witch of this phones or another ones gave you the best results in a productivity enviroment? Thanks in

[asterisk-users] Activate/Deactivate zap channels in realtime

Hi all, I am looking for a solution for the following problem. I have a little callcenter with 20 agents and 20 incomming analog lines, one for each agent. I need to have abailable as incomming analog lines (FXO Ports) as agents logged, not all the agents are logged all the time. It is needed

[asterisk-users] Test to Speech

Hello all, I am looking for software for text to speech in spanish witch works with asterisk (1.2.13). I have tested festival and the cepstral software, both works but the quality is so poor in the spanish language. Someone has worked with any test to speech software with aceptable quality in

[asterisk-users] snmp Monitor for asterisk boxes

asterisk based on your experience? Thanks a lot Voip Crazy. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users