[Asterisk-Users] Asttapi
Hello all, Just to inform you all, next version released, please try it and let me know about any bugs you find (or any further features). This release now includes 1/ Inbound calls 2/ Call origination 3/ Call dialling from phone detected 4/ Call origination using contexts 5/ Can set the caller ID 6/ Priority field added to the origination Some bugs fixed also. Known bug at the moment - Caller ID number not presented correctly to TAPI - next to work on. Download from http://www.sf.net/projects/asttapi documentation at www.omniis.com/asttapi Regards Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco phones
Hello all, I havent used cisco phones as yet do they work with asterisk, are they good which models are the best? I am after a starting point! Thanks Nick
[Asterisk-Users] Asttapi
Hello all, Just to update, Instruction's can be found at www.omniis.com/asttapi, including where to download it from. This is update 0.02, this now includes a little feedback from Asterisk so that when click to dial has occurred then it is indicated at the start and the end of the call. Now working on inbound calls. Any question, please send to me. Regards Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] reboots
Hello all, Just looking for some opinions. What is the expected uptime for asterisk assuming the box has all the resources it needs. I ask this because I have only to date seen max 9 days which appears very low. This is a system only running Asterisk. It has 1.5GB RAM with 2GHz processor, there are 8 users although not always simultaneously it is a fairly well used system. With a traditional phone system you would expect to power it up and just leave it, so what about Asterisk con job for a reboot? Regards Nick
[Asterisk-Users] Recall: reboots
Title: Recall: reboots Nick Knight would like to recall the message, reboots.
[Asterisk-Users] reboots
Hello all, Just looking for some opinions. What is the expected uptime for asterisk - assuming the box has all the resources it needs. I ask this because I have only to date seen max 9 days which appears very low. This is a system only running Asterisk. It has 1.5GB RAM with 2GHz processor, there are 8 users - although not always simultaneously - it is a fairly well used system. With a traditional phone system you would expect to power it up and just leave it, so what about Asterisk - con job for a reboot? Regards Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] reboots
The kernel is 2.4.22 It is a gentoo box, although it had a vanilla kernel installed, CAPI was patched into the kernel for using CAPI drivers. It uses Asterisk version 1 from CVS, running SIP clients for the phones and CAPI across an eicon diva card (4bri). cacofonix root # uname -a Linux cacofonix 2.4.24 #5 Sun Apr 4 13:54:33 GMT 2004 i686 Intel(R) Celeron(R) CPU 2.00GHz GenuineIntel GNU/Linux cacofonix root # free total used free sharedbuffers cached Mem:514408 509424 4984 0 65880 300652 -/+ buffers/cache: 142892 371516 Swap: 1004052 01004052 cacofonix root # it uses kapjods rtc plugin, and runs MOH. It queues calls and runs some mailboxes. We have 7 users in the office, there is a good chance there is 3 calls on the go at any one time. Let me know if you need any other information I am going to go to kapejods 4 bri card with kernel 2.6 - but I am unsure wether this will follow me! Thanks Nick -Original Message- From: James H. Cloos Jr. [mailto:[EMAIL PROTECTED] Sent: 20 April 2004 13:43 To: Nick Knight Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] reboots Nick == Nick Knight [EMAIL PROTECTED] writes: Nick What is the expected uptime for asterisk - assuming the Nick box has all the resources it needs. Months. You should only have to reboot for kernel updates and restart * when updating it or (some parts of) its configuration. Nick I ask this because I have only to date seen max 9 days Nick which appears very low. Something is definitely wrong with that box. To diagnose will require at least details on processor, kernel version, distribution, asterisk version, and hardware installed. -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TAPI driver
Hello all, Just a quick note, I have been putting together a TAPI driver for Asterisk, this enables the user to perform things like click to dial from any TAPI enabled app (such as outlook or ACT etc). At the moment it is very basic and can only perform click to dial but further functionality will be coming. It uses the Asterisk manager to place calls. Please feel free to use it - not much documentation as yet but will be coming, can be found on sourceforge project name asttapi. Regards Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TAPI driver
Hello all, Just a quick note, I have been putting together a TAPI driver for Asterisk, this enables the user to perform things like click to dial from any TAPI enabled app (such as outlook or ACT etc). At the moment it is very basic and can only perform click to dial but further functionality will be coming. It uses the Asterisk manager to place calls. Please feel free to use it not much documentation as yet but will be coming, can be found on sourceforge project name asttapi. Regards Nick
[Asterisk-Users] SIP or any softphone on Mac os x
Hello all, I have just made a switch to running MAC os x as my desktop, I was using Xten and in the past pingtel softpones on my computer, but now need to find a new product for the MAC. I tried the xten, but it appears to be very flakey - has any one any experience of softphone on MAC? Thanks Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] firefly softphone
Hello all, I have tried the firefly softphone on a couple of computers now and as soon as it registers with the Asterisk server (in fact tries to register) but then crashes and tries to send crash report to MS. Has any one had experience of this. Nick
[Asterisk-Users] differentiate incoming calls on SIP clients
Hello all, I would like to set-up some direct lines so that when a user of mine answers the phone he/she knows to say the correct intro message, so that we can introduce ourselves as different companies. I have played around with caller ID and can modify that using caller ID Name doesnt seem to work unless it is numeric? Do you have any imaginative solutions for this? Thanks Nick
[Asterisk-Users] SIP clipping sound
Hello all, Is there a way of setting the sound level at which * starts to transmit silence. It appears when an external call comes in the caller speaks silently you hear a lot of lost bits as it drops in and out. This only seems to have been introduced when I upgraded to the latest version of *. Regards Nick
[Asterisk-Users] agents and call queueing
Hello, I have been playing around with call queuing very cool. So at the same time I also tried to implement the agent via the agent call back routine. This is causing problems, in the queue.conf if I have a member as Member = Sip/nick It works But if I set up an agent, login using AgentCallBackLogin, the login works, and when a call is entered into the queue then the phone rings but as soon as the sip phone picks up the call then the call is dropped and the call is returned to the queue (whilst listening to on hold music!). Help Thanks Nick
[Asterisk-Users] asterisk sip with voicemail
Hello all, I have setup my sip.conf so users can register etc in the following format, [person] type=friend username=nick secret= host=dynamic mailbox=101 in my voicemail.conf I have an entry like 101 = 1234,Nick Knight,[EMAIL PROTECTED] Leaving a voicemail works fine after I have my dial command time out but on sip clients which display whether voicemail is waiting or not, it always displays No Voicemail (pingtel expressa). It emails them fine and I can call into the voicemail app which says I have new voicemail but I would like to resolve the sip client issue. Thanks Nick
[Asterisk-Users] re: asterisk sip with voicemail
Hello again, I have tried removing the username the username and the [] where the same just didnt copy it that accuratelyJ and the sip clients appears to register still ok but it is still doesnt register the fact there is voicemail waiting. Nick i guess the user * looks for is the text within [] so i suppose the = username and the text within [] should be same. try putting [nick] in = place of [person]. actually, u don't need username as it only looks for = "" text between [] as username.
[Asterisk-Users] call files
I am after using a web crm system which has a button to then get asterisk to dial the contact. For this I was looking at call files, which appear good for the job, I have one small problem with them though. 1/ file is created 2/ external number is called 3/ the external party answers 4/ the external party now hears ringing as you extension is now being called bad! What I would like to get round this is probably the reverse I dont want the people I am calling to hear ringing. For example as soon as it has dialled the receiving end call me, or call me first then call the other extension? It is probably something very simple I am missing! Nick
[Asterisk-Users] windows messenger and DTMF
Hello All, Another question todayJ I have just started playing with windows messenger and asterisk following the little how-to from the asterisk web-site work well good sound quality but you cannot put people on hold transfer them or send DTMF (to get asterisk to do the transfer) or am I missing something this makes quite a poor soft phone if this is the case? Thanks Nick
[Asterisk-Users] Re: call files
I am after using a web crm system which has a button to then get asterisk to dial the contact. For this I was looking at call files, which appear good for the job, I have one small problem with them though. 1/ file is created 2/ external number is called 3/ the external party answers 4/ the external party now hears ringing as you extension is now being called bad! What I would like to get round this is probably the reverse I dont want the people I am calling to hear ringing. For example as soon as it has dialled the receiving end call me, or call me first then call the other extension? It is probably something very simple I am missing! Swap the numbers around. Hello again, I cannot figure this out - just swap them round? I have Channel: CAPI/isdn number::number MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: internal Extension: 101 Priority: 1 But if I swap it round Channel: SIP/User MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: internal Extension: ??? Priority: 1 I have tried various things on the extension but cannot figure out how that would works up to that point as it calls my extension but then obviously fails when it cannot figure out what to do on the return??? Thanks again Nick
RE: [Asterisk-Users] Re: call files
I have tried this from the manager console and call files and it doesn't
seem to work the other way round. It will call the sip channel but not
the capi channel - in fact with capi debug this doesn't show anything
getting through
Asterisk monitor comes up twith
Attempting call on sip/nick ofr number@isdnout:1 (Retry 1)
Channel Sip/nick-dd98 was answered.
Then the sip call is dropped (by asterisk) then nothing!
Any others ideas???
Nick
-Original Message-
From: Philipp von Klitzing
[mailto:[EMAIL PROTECTED]
Sent: 22 December 2003 18:17
To: Nick Knight
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: call files
Hi!
What I would like to get round this is probably the reverse I
don(tm)t
want the people I am calling to hear ringing. For example as soon as
it
Swap the numbers around.
I cannot figure this out - just swap them round?
But if I swap it round
Channel: SIP/User
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: internal
Extension: ???
Priority: 1
Try something like this:
Context: isdn_outgoing
Extension: 12345678
Priority: 1
[isdn_outgoing]
exten = .,1,Dial(CAPI/yourMSN:${EXTEN},,rT)
Cheers, Philipp
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Re: [Asterisk-Users] music on hold (SIP Clients)
Thanks for that - but how does that plug into a sip client - all this will do as I understand it is if I forward a call to that extension it will play music - how do I get it back - and how do I tie it into the hold button on a sip client?? Thanks again. Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi
Hello All, I have been using chan_capi with eicon diva server 4 Bri for a few months now and been very happy with the results. I have now got a small problem. I needed to upgrade - get more lines in - during an office move. So we moved office and landed with 3 ISDN2e lines. I can place calls perfectly. Problem is Asterisk is only receiving calls on 1 channel - so effectively I am only getting 1 out of 3 calls (BT ran some protocol tests on the line). The ISDN lines are setup to all share the same number. I have adjusted capi.conf from ; ; CAPI config ; ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] ;isdnmode=ptp ; ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] ;isdnmode=ptp msn=870582 incomingmsn=870582 ;incomingmsn=870582,870583,870878,871310,871743,871821,872215,872355 ;incomingmsn=* controller=2 context=capiIN devices=2 To: ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] ;isdnmode=ptp ; ; CAPI config ; ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] ;isdnmode=ptp msn=870582 incomingmsn=870582 ;incomingmsn=870582,870583,870878,871310,871743,871821,872215,872355 ;incomingmsn=* controller=1,2,3 context=capiIN devices=6 the asterisk startup script is: ; #!/bin/bash case $1 in stop) killall asterisk rmmod capi rmmod divacapi rmmod divas ;; start) modprobe divas modprobe divacapi modprobe capi divactrl load -c 1 -f ETSI -u -t 0 asterisk ;; I have tried various computations of the config - but don't really understand it - can someone help please Regards Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: USB handsets/headsets?? (WipeOut)
Hello Headsets we use are Plantronic's which are great, but I am still looking for a USB handset. Regards Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] USB handsets/headsets??
Voipvoice handsets we tried - and are now sat on a shelve gathering dust. The main problem was the quality of the audio - to quiet and poor - not telephony grade for the office - perhaps good enough for home use. Just my two pennys! But still looking for a usb handset! Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] csv phone log
Hello all, I am thinking of giving my employees incentives - I want them to speak with my clients more (on the phone) - and in order to do so I need to measure it! I have had a look through the master csv file which asterisk produces - but I am not sure how to interpret it. I am looking for 1/ How long each employee is on the phone for 2/ How many calls each employee answers 3/ Who they are speaking with 4/ Are we receiving more or are we making more etc 4/ and so on... I have crystal reports ready to rip the file apart but I am unsure what all the fields are for. Any pointers would be appreciated. Thanks Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and pingtel
Hello All, I have pingtel and asterisk working really well. I have a really annoying little problem - mainly with pingtel. When a call comes in pingtel displays the caller ID on the phone. If I miss it then I click on the number for redial - this doesn't include a 9 to dial an outside line. The second problem is with the dialer from outlook again it bypasses the outlook dialing rules so doesn't include a 9 for an outside line. Does anyone have experience of these problems and imaginative ways round (or even fixes!) Regards Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] QOS
Hello all, Apologies as not really an Asterisk question - QOS. I have been told to implement VOIP correctly you need QOS implemented across the network as a whole. What network switches support this? Regards Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] monitor
Hello all again, Last time you helped by suggestiong that monitor will record by telephone conversations - I have added this to my config - but where does it save the files? Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] monitor
no /var/spool/asterisk/monitor - although there is a /var/spool/asterisk
clippnig from extensions.conf
exten = 870582,1,Wait(1)
exten = 870582,2,Monitor()
exten = 870582,3,Dial(${EVERYONE},10)
exten = 870582,4,capiCD(${NUMBER})
exten = 870582,5,Hangup()
Nick
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[Asterisk-Users] pingtel phones
Hello all, Hope I am not too of topic here - but it cross's the phone/asterisk boundary. I have been playing with a few soft phones - noticed that pingtel seemed to be highly recommended across previous postings. I have been using xten - which is a great phone but seems a bit limited in its functionality - which is why I am now looking at pingtel. Problem is I cannot get it to register properly against asterisk - I am know at a bit of a loss - can someone point me/let me have a simple howto to get asterisk working with pingtel softphone. BTW they work in an unregistered mode - but obviously need the registering part so asterisk can place a call back to the softphone. Thanks Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF onto the reall world
Hello All, I was originally having problems just getting DTM from the SIP phone over to asterisk for the voice mail app. But through settings in the SIP client got that working. I am know at the stage know though that asterisk doesn't appear to be passing DTMF over to the real world, I am using ISDN 4 Linux with an ISDN2e. What settings in Asterisk are responsible for this? Thanks Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] isdn4linux
Hello All, I was just after a few pointers. I have just setup my Redhat 9 linux box with asterisk. Internal SIP call working fine. My Eicon ISDN card turned up today so - plugged it in and went through the modem.conf. It reports unable to open /dev/ttyI0 The problem is I have never used ISDN with Linux - let alone a telephony app - and I have no idea even where to start. Some pointers would be appreciated. Regards Nick Ë^®+$R²f¢)à+-Ë^®+$R²X¬¶Çb+¦r¡¶ÚþX¬¶Çb+¦r¿¨¥©ÿ+-wèý«-z¸¬ë ®
[Asterisk-Users] Phones
Hello all, I am a newbie to this list - and so far very impressed with the functionality of Asterisk. So far I have setup a simple soft phone running on a windows PC making calls to other SIP soft phones. Later this week I hope to get UK ISDN2e up and running with it! My question is I would like the experience and feedback from users about what equipment/software you are all using for phones to connect to Asterisk, so fat I have been playing with xten soft phone which works very well but before I make my decision on which phones to use would like the feedback of the group. Regards Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
