[Asterisk-Users] DeadAGI Application
I downloaded the astcc calling card program. Thanks, it is very easy to setup and works Excellent. Anyway, it says to use DeadAGI to run it rather than AGI. I don't know what I am doing wrong. I just updated my asterisk from cvs and rebuilt and reinstalled. I do not have an application called DeadAGI. I have searched the source, google, etc. but have not been able to find anything. Any pointers? Darren Wiebe [EMAIL PROTECTED] ___ 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] Jitter over Sat
On Tue, 31 Aug 2004, Storm D. J. Petersen wrote: > I have a problem with jitter over a 2mb up 1mb down satellite connection. I > call my friend over the satellite - I call perfect but they cannot make out > a word I say. However if I leave him voicemail on his asterisk box, it > records my voice perfect. I have this problem when calling other people as > well. It sounds like you just don't have enough throughput in the one direction. Voicemail is fine because it doesn't need "realtime" capacty - the voice frames arriving from your side go into the captured file as they arrive, doesn't matter if your 10 second message takes 20 seconds to arrive... You could try using a lower bandwidth codec like GSM if you aren't already. Steve ___ 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] iaxtel and jitterbuffer
On Sun, 29 Aug 2004, Kris Boutilier wrote: > Is timestamp information calculated purely from the relative timestamps of > each frame of the current incoming stream or is there some degree of RTC > synchronization expected between the two endpoints? No sync is needed; its all relative. > Similarly, are jitter calculations made seperately for each discrete channel > (ie. the IAX level) or are they based on an aggregate of all channels > between each pair of two endpoints (ie. the TCP/IP level)? De-jtter is done for each call independently. Steve ___ 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] Line death not recognized on TDM400P?
On 1 Sep 2004 at 17:15, [EMAIL PROTECTED] wrote: > A customer of mine has 3 TDM400P cards in a box running asterisk. On > each card he has four FXO modules. > > I have set up the dialplan to dial via group 1 for an outgoing call. > > Channels 1-12 are in group 1. > > If he plugs a telephone cable into socket 2 or 3 etc, but not 1, when > he dials out, it still tries to make the call via socket 1. > > Straight away the console says that it has dialed the number via g1 > and that it is connecting sip/bla with zap/1-1 (or some such)... > > On my X100P I get a red alarm if the phone cable is not plugged in. > Is there any way to do this with the TDM400P? > > They would like to be able to unplug lines and use them for other > purposes at times. > > Make sense? > > I kinda thought that asterisk would realise that nothing was > connected to the TDM card and try the second socket, the third etc... > > Any help greatly appreciated. > The problem is, if lines 1,3,4,5,6,7,8,9,10,11 and 12 are plugged in, you would only be able to make 1 concurrent call...(because the next call would try to go out line two which would never work)...maybe if four people called at the same time only 1 wouldn't get through but then the next call wouldn't get through... Is this so? Matt ___ 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] Dial/Zap doesn't work
Hi, I'm trying to dial in from one phone and give it access to another line (ie incoming on zap/1 and outgoing on zap/2)... how can I transfer the call from channel 1 and give it the dial tone on channel 2? I can use dial but that takes a phone number, which I want the user to be able to select. thanks Imran ___ 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] Line death not recognized on TDM400P?
A customer of mine has 3 TDM400P cards in a box running asterisk. On each card he has four FXO modules. I have set up the dialplan to dial via group 1 for an outgoing call. Channels 1-12 are in group 1. If he plugs a telephone cable into socket 2 or 3 etc, but not 1, when he dials out, it still tries to make the call via socket 1. Straight away the console says that it has dialed the number via g1 and that it is connecting sip/bla with zap/1-1 (or some such)... On my X100P I get a red alarm if the phone cable is not plugged in. Is there any way to do this with the TDM400P? They would like to be able to unplug lines and use them for other purposes at times. Make sense? I kinda thought that asterisk would realise that nothing was connected to the TDM card and try the second socket, the third etc... Any help greatly appreciated. Cheers, Matt Riddell ___ 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] OT: Headset for Cisco 7960?
On Tue, 31 Aug 2004 15:58:16 -0500, B. J. Bomar <[EMAIL PROTECTED]> wrote: > I use a Plantronics Supra H51 plugged straight into the headset port, and it > works great. > > B. J. Same here. They're wonderful headsets. -Shaun ___ 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] Polycom SoundPoint IP 300 Configuration
Look up the word persist in the XML config file... - Brent On Tue, 31 Aug 2004, Reid A. Forrest wrote: > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Matthew Marlowe > > Sent: Monday, August 30, 2004 12:55 PM > > To: [EMAIL PROTECTED] > > Subject: [Asterisk-Users] Polycom SoundPoint IP 300 Configuration > > > > I just got a Polycom soundpoint and I set it up using the > > phone and web > > based admin. > > > > I cant seem to figure out the config files and they are confusing me > > greatly and I dont have time for it :) > > > > Some things are odd, like on every reboot it seems the volume I set is > > reset? is there any way to fix that. And the ringer seems low. - Even > > all the way up > > > > The volume reset is intentional on Polycom's partt, due to US ADA > restrictions (Americans with Disabilities Act). It must reset after each > call. This can be overridden through the config files, althrough I can't > recall the exact setting right now. Email me off list if you can't find it > and I'll look it up. > ___ > 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 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] Does anyone have a working GR-303 config?
Chris Jensen wrote: I am hooking up to a DMS500 (100&250 together) and wanted to see if anyone had any experience with this. We have the GR-303 span up, the IDT is built. I have not yet heard of anyone doing this, but would be _extremely_ interested in your experiences. Please keep in touch with me off-list, if you would. Thanks. ___ 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] install software version to mediatrix 1204 (how to)
i'm new here and i need help on how where can i get software version 4.0.x of the mediatrix and how can i install it... mediatrix unit im using has a software version of 2.4.9.57. i would like to use H.323 not SIP... please need help asap!... hope to hear from anyone of you soon.. thanks in advance! -- eder ___ Do you Yahoo!? Win 1 of 4,000 free domain names from Yahoo! Enter now. http://promotions.yahoo.com/goldrush ___ 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] All you polycom folks.....
Just out of curiosity, What version of CVS and Polycom SIP software are you running happily? Are you running SIP 2.3.0 yet? 2.2.0? 2.1.0? I tried upgrading the CVS yesterday, with a mixed mode of 2.2 and 2.1 with poor results. Transferring did not work as expected. Using the # key to do blind transfers after a call was on hold did not work. Just curious. Thanks, - Brent ___ 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] No audio on PRI channel answered by Playback()orMeetMe()
Tobias Jönsson wrote: Sorry, I did not know these american specialities. I just noticed in Larry's PRI debug info that he received a STATUS message during the waiting, so I thought that the waiting could lead to some kind of timeout at the telco. In EuroISDN the callerid always come in first SETUP message and so it did in Larry's pri debug. The calling number _is_ delivered in the SETUP message; what is not delivered (in National ISDN-2) is the calling name. That comes later in a FACILITY message, and if you Dial() an extension before it has arrived, the destination phone won't see the calling name. ___ 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] PLC (Packet loss cancel) questions
On Aug 31, 2004, at 8:42 AM, Steve Underwood wrote: Chris Shaw wrote: - Channel Support: IAX2 in asterisk IAX2 in libiax2 Other IP channels in asterisk (RTP-based ones, I guess are all that is left). CNG/VAD and DTX in SIP is a must if * is to be taken seriously as a complete solution... As much as we all hate it's complexity and wish that everything would speak IAX (I know I do) a large number of devices support (and will be supporting) SIP, making it equally as important as IAX2 in using * as a complete telephony solution... This is nothing to do with SIP. It is an RTP issue, common to everything which uses RTP - SIP and H.323 included. Sending no packets is perfectly valid, and normal, in RTP. If the receiving end takes no packets (other than, perhaps, an extremely long silence) as a disconnect it does not comply with the RTP spec. DTX is much despised, and CNG only slightly better. They just sound good (pun intende) on paper. They make a big difference in conferences, actually; In a big conference, you generally have just zero to 2 speakers at any time, and many many calls that are not transmitting. In this case, it is wasteful of all kinds of resources for these clients to transmit anything. Not just bandwidth, but also processing in your mixer. We currently have deployed app_conference (sources with iaxclient), which depends on iaxclient's ability to do DTX transmission in order to be scalable. DTX Support: Sending a single CN packet (in IAX2, this should probably sent reliably) would probably be good. I second, third and fourth this one as does anyone who's tried to use BroadVoice with Voicemail... Currently when * is not making any noise (e.g. recording) absolutely NO packets are sent back to the proxy... A lot of proxies take this as a sign that the far end has disconnected... Including BroadWorks! But they do recognize small CN packets as a sign that the SIP device (Asterisk) is still there... A lot of CNG spec. call for only one transmission, and then silence. Continued CNG has real benefits, but it certainly not the norm. PLC I think is somewhat implemented already in codecs that support it, but I could be wrong, I remember seeing mention of it in the code... PLC is seldom included in the codecs. If you read the specs they often mention PLC, but only in terms of how the codec mitigates the awfulness of a lost packet. Few codecs actually include it. iLBC and speex both include it. Speex' PLC is triggered by simply passing in NULL instead of a pointer to compressed audio. It will then generate a predicted frame without any information: I imagine it basically continues the energy transmission of previous packets. app_conference currently emulates PLC for GSM or by sending the GSM decoder the previous packet (it only does this if it's sucessfully gotten a few packets in a row). This isn't just something that looks good on paper, but something that significantly improves the quality of conversations in practice. (although, in this practice, the lost packets are just as often artifacts of scheduling issues and the present jitter buffer as they are from actual network loss). PLC is also really important when using a dynamic jitter buffer, because whenever the jitter buffer wants to grow, PLC can hide the sparse frames. While the current jitter buffer will just give you a big gap, ideally, if you're growing, you'd be able to spread the gaps a bit, and use PLC to hide them. For example, if you have 100ms worth of frames already queued, and you need to grow the jitterbuffer to 180ms, you'd intersperse the 80ms gaps into 20ms (or whatever) size gaps, and use PLC to interpolate them. This would be much less noticable than an 80ms dropout in audio. -SteveK ___ 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] T100P Configuration for Mixed Voice & Data
I need to know how to setup the data side of the T1 on my Linux Box. I have found information about configuring a PRI and HDLC but nothing about the Frame-Relay type setup for data. The following is information from our T1 provider. Network T1: Framing = ESF Line code = B8ZS Build out = 0-133ft(DSX)/0dB(CSU) Clock = network Pulse-density-enforce = off alarm-option = on alarm-delay = 15 is-slave = off DS0 Provisioning: analog-begin = 1 analog-end = 16 data-begin = 17 data-end = 24 alignment = same (The following is what our Vina T1 Integrator currently has in its settings. Our Linux box will replace the Vina) Synchronous Interface: encapsulation = Frame-Relay HDLC-inversion = off Encap-data: LMI-Type = T1.617-annex-D N391 = 6 N392 = 3 N393 = 3 T391 = 6 PVC1 : DLCI = 100 IP = ---.---.---.--- netmask = 255.255.255.252 RIP = disabled NAT = OUT Thank you for the help ahead of time! --Shawn ___ 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] Analog lines and TDM card
Correct. TDM (time division multiplex) FXO is for analog ports coming from the telco. - Original Message - From: "Marcello Lupo" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, August 31, 2004 1:52 PM Subject: [Asterisk-Users] Analog lines and TDM card > Hi, > sorry to bother you, but i need to connect 8 standard analog lines to 2 > asterisk servers (one in Italy (4 lines) and one in USA (4 lines)) and after > let this 2 systems to interact between them. > I was thinking to use the TDM400 card equipped with 4 FXO modules on both > sides. > Is it correct to do this (use the TDM card to terminate analog lines) or i > have to use 4 X100P PCI card in both servers? > Thank you in advance. > Bye, > Marcello > > > ___ > 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 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] Losing voice on Digium demo server - how to spotproblem ?
try steven sokol's iaxphone and see if you have the same problems dialing
his box while taking * out of the equation. same problem=network, no
problem = *
http://www.sokol-associates.com/IaxPhoneDownload.htm
- Original Message -
From: "Robert Rozman" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Tuesday, August 31, 2004 1:51 PM
Subject: [Asterisk-Users] Losing voice on Digium demo server - how to
spotproblem ?
> Hi,
>
> I'm trying to get Asterisk working on P4 2.8 server behind NAT and
Firewall
> (all ports we're set according to instructions) on DSL line.
>
> When pbx connects to Digium demo server( I'm located in Slovenia, Europe),
> it gets first few words, then silence and then comes back when enumerating
> dial possibilities ("4 for accounting ...). Same happens from SIP or IAX
> local extension.
>
> I guess this is network problem, but would kindly ask for guidance for
what
> measures should I take and what seetings are first to try to avoid this
> problems. I have another server running at my home on dialup line
(28.8kbps)
> and it connects to digium without problems, so I'm little suspicious being
> only network traffic problem.
>
> Thanks in advance for your effort,
>
> regards,
>
> Robert.
>
> ___
> 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 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] Polycom SoundPoint IP 300 Configuration
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Matthew Marlowe > Sent: Monday, August 30, 2004 12:55 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Polycom SoundPoint IP 300 Configuration > > I just got a Polycom soundpoint and I set it up using the > phone and web > based admin. > > I cant seem to figure out the config files and they are confusing me > greatly and I dont have time for it :) > > Some things are odd, like on every reboot it seems the volume I set is > reset? is there any way to fix that. And the ringer seems low. - Even > all the way up > The volume reset is intentional on Polycom's partt, due to US ADA restrictions (Americans with Disabilities Act). It must reset after each call. This can be overridden through the config files, althrough I can't recall the exact setting right now. Email me off list if you can't find it and I'll look it up. ___ 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] SIP registration with public dynamic ip address
I set up my own STUN server and turned reinvite off. Lyle - Original Message - From: [EMAIL PROTECTED] To: '[EMAIL PROTECTED]' Sent: Tuesday, August 31, 2004 8:53 AM Subject: [Asterisk-Users] SIP registration with public dynamic ip address Hi, I'm trying to configure a natted budgetone phone to a asterisk server as described in wiki using port forwarding. I successfully make call from the client but it seems it does not register the client ip address and when I try to recall it is not reacheable. Asterisk can manage natted sip client with dynamic ip address ? Bye ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] SMS & Asterisk - an explanation
Maxim- This will not work through a FWD DID as you suggest. BT requires each telephone number to be registered in order to receive SMS messages. You need a either an analogue, BRI, or PRI line that terminates in your asterisk box directly. The way a line gets registered is that you must initiate a special SMS message on the asterisk box from the number you are registering (see the details on the asterisk Wiki page, under the SMS command) Once you have registered a number, you can send an SMS text message from a UK mobile to that number, and the asterisk box will receive it, assuming that you've defined SMS handling for that number in your dialplan. IMPORTANT NOTE: As of two weeks ago when I tested this, BT is only accepting SMS's from Vodafone mobiles - O2 and Orange do not work yet. (not sure about T-Mobile) You can send messages to all carriers however. This is expected to change in the next couple months when BT will accept SMS's from all carriers. Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California & London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxim Litnitsky Sent: Tuesday, August 31, 2004 5:19 PM calluk.com gives free 0870 DIDs. I registered my 0870 to FWD account, and FWD passes all to my * box. When I send SMS to my 0870 DID, * shows nothing and I get SMS error. ___ ___ 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: AW: AW: [Asterisk-Users] SMS & Asterisk
On Tue, 31 Aug 2004 15:34:51 +0200, Axel Eble <[EMAIL PROTECTED]> wrote: > On Tue, 31 Aug 2004 15:22:26 +0200, Michael Labuschke > <[EMAIL PROTECTED]> wrote: > > > > > > Pick up mobile phone.. enter sms .. send it to the * phone number. > > Done > > On the * side.. follow the sms howto (voip-info.org might have some infos) > > > > Done > > Ah. That requires SMS to be available on land lines. > > > > Axel > > -- > Axel Eble, CISSP * Trienter Str. 6b * 87437 Kempten (Allgäu) * Germany > VoIP: [EMAIL PROTECTED] * cell: +49.178.285-3265 > ___ > 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 > calluk.com gives free 0870 DIDs. I registered my 0870 to FWD account, and FWD passes all to my * box. When I send SMS to my 0870 DID, * shows nothing and I get SMS error. ___ 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] Asterisk codecs and packet size
Luis Vazquez [EMAIL PROTECTED] wrote: > Does anybody knows if it's posible or if there is some develoment in > course to be able to use longer transmit packet sizes (as long as I know > this is fixed in 20ms now) with the compressed voip codecs in asterisk > (g729, g726, gsm, etc). I need to use asterisk to connect remote sip > clients with 24kb bandwidth lines and I'm using a licences g729 codec but > because I can't increase the packet size to 40 or 60 ms in asterisk the > connection is useless. Thanks very much Luis > It wouldn't help you if there was an easy Asterisk patch for this, as the G.729 code is closed source and is therefore un-patchable. You could try SpeeX or LPC10 - or a 56k modem. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] MP3Player strange error
Hi all!
I downloaded right mpg123, chabged path to mpg123 binary in
app_mp3.c, rebuilt app_mp3.so, and got MusicOnHold to work. But
MP3Player refuses to do properly:
-- Accepting AUTHENTICATED call from x.x.x.x, requested format =
1024, actual format = 1024
-- Executing Answer("IAX2/[EMAIL PROTECTED]/3", "") in new stack
-- Executing MP3Player("IAX2/[EMAIL PROTECTED]/3", "mohmp3") in new stack
Aug 31 08:07:25 NOTICE[1135618864]: chan_iax2.c:2375 iax2_read: I
should never be called!
Aug 31 08:07:28 NOTICE[1135618864]: app_mp3.c:91 timed_read: Selected
timed out/errored out with 0
-- Timeout on IAX2/[EMAIL PROTECTED]/3
== CDR updated on IAX2/[EMAIL PROTECTED]/3
-- Executing Hangup("IAX2/[EMAIL PROTECTED]/3", "") in new stack
== Spawn extension (litnimax-in, t, 1) exited non-zero on
'IAX2/[EMAIL PROTECTED]/3'
-- Executing Hangup("IAX2/[EMAIL PROTECTED]/3", "") in new stack
== Spawn extension (litnimax-in, h, 1) exited non-zero on
'IAX2/[EMAIL PROTECTED]/3'
-- Hungup 'IAX2/[EMAIL PROTECTED]/3'
new*CLI>
___
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 79XX SIP Ring Tones
Hi all, Has anyone gotten custom ring tones to work using ALERT_INFO with the Cisco 7940 SIP phone? I've read the wiki, but just can't get this to work. I'm currently using the 7.2 SIP image. Thanks, Chris ___ 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] Zap & ANSWER the Call
The standard for loop start does not send answer supervision, so * and all
other telcom devices that do CDR records have to 'assume' that the call was
answered.
Lyle
- Original Message -
From: "Rodrigo P. Telles" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Tuesday, August 31, 2004 5:31 PM
Subject: Re: [Asterisk-Users] Zap & ANSWER the Call
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi,
>
> Nobody knows about that strange "behaviour" of Zap channels or
> at least if is that right?
>
> Thanks in advance.
>
> Rodrigo P. Telles wrote:
> | Hi,
> |
> | I'm using a TDM400 with one FXS and one FXO module (developer kit) and
> | I've been testing termination from SIP phones to PSTN and it works fine,
> | but
> | asterisk accounting is doing something strange (for me).
> | Scenario:
> | 1 - extension 1009 (SIP phone - BT101)
> | 2 - Zap/4-1 (TDM400 FXO module)
> |
> | extensions.conf:
> | [dialout]
> | exten => _9.,1,Dial(Zap/4/${EXTEN:1}|25|r)
> | exten => _9.,2,Congestion
> |
> | [sip]
> | include => dialout
> | exten => 1009,1,Dial(SIP/1009,20,rt)
> |
> | So, when I dial "9something" from 1009, something rings and then I
> | hangup the phone.
> | I realised that asterisk thought that "something" ANSWERED the call:
> |
> |
"","1009","9something","sip","""Tests""","SIP/1009-42fb","Zap/4-1","Dial",
> | "Zap/4/something|25","2004-08-27 20:15:34","200
> | 4-08-27 20:15:36","2004-08-27 20:15:43",9,7,"ANSWERED","BILLING"
> |
> | Is that right?
> |
> | Version: Asterisk 0.9.0
> |
> | Thanks in advance.
> |
> | --
> |
> | Rodrigo P. Telles <[EMAIL PROTECTED]>
> | Project Manager
> | Devel-IT - http://www.devel-it.com.br
> | TDKOM Group
> |
> ___
> 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
>
> - --
>
> Rodrigo P. Telles <[EMAIL PROTECTED]>
> Project Manager
> Devel-IT - http://www.devel-it.com.br
> TDKOM Group
>
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.0.7 (GNU/Linux)
>
> iD8DBQFBNPxViLK8unYgEMQRAnJ1AJ0WLPUHTMW9XJif5y5iECJJoloU8QCdHp1G
> TGwaMVemCOPE1uOZQPFKdnc=
> =zq37
> -END PGP SIGNATURE-
> ___
> 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 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] Asterisk codecs and packet size
Does anybody knows if it's posible or if there is some develoment in course to be able to use longer transmit packet sizes (as long as I know this is fixed in 20ms now) with the compressed voip codecs in asterisk (g729, g726, gsm, etc). I need to use asterisk to connect remote sip clients with 24kb bandwidth lines and I'm using a licences g729 codec but because I can't increase the packet size to 40 or 60 ms in asterisk the connection is useless. Thanks very much Luis ___ 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] good Dutch TTS ?
hi; anyone can recommend a good TTS for the dutch language compat in linux? -- Best regards, Danny mailto:[EMAIL PROTECTED] belGOnet.com a Euro-pictures division - internet solutions place princesse elisabeth 9/11 - 1030 Brussels - Belgium Tel : +32-(0)2-215.67.65 - Fax : +32-(0)2-215.66.65 domains - hosting - hardware - VoiP - consultancy - backuping CISCO - HP/COMPAQ - SUN - EMC - JUNIPER - IBM - DELL - NORTEL No legal consequences can be derived from the contents of the email neither is belGOnet.com committed to them. The content of this email is exclusively intended for adressee(s) and information purposes. belGOnet.com accepts no liability for any damage resulting from the use and/or acceptation of the content of this email. ___ 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] Can only call asterisk once
I have Asterisk RC2 setup here with a Fritz ISDN card on Debian Woody. I'm using chan_capi-0.3.5 and fcpci-suse8.2-03.11.02. Settings are pretty much the default, except in order to get the fcpci module to compile, I had to follow the instructions here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20AVM%20Fritz%20CAPI%20Driver%20Install but the drivers would only compile if I left the CCFLAGS as they were. Now I've been able to successfully call Asterisk from a POTS phone. We have a block of 10 numbers that are on the ISDN line. Once I call one of those numbers, no more calls will go through after that. I have to restart Asterisk in order for it to answer another call. Has anyone seen this behaviour before? It's certainly not ideal ;) Thanks -- James Doherty Zeald.com Network Operations Ph: +64 9 415 7575, Fax: +64 9 443 9794 Web: http://www.zeald.com ___ 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] Newbie - Voicemail Password Help
The vm-password file is used else where, such as queues. If you changed
it, then you would change for all the other applications.
I guess, since we alter the source code often.. it's not that big of a
deal. We just create our own patch files and if we update from cvs, we
patch against the new source.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Umar Sear
Sent: Tuesday, August 31, 2004 12:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Newbie - Voicemail Password Help
On Tue, 2004-08-31 at 10:53, Stephen Hon wrote:
> Paul,
>
> What you can do is modify the source code for the voicemail
application.
>
> Edit line 3579 in /usr/src/asterisk/apps/app_voicemail.c. Change the
file 'vm-password' to 'pls-enter-vm-password'.
>
> Recompile and install.
>
> Then in your macro remove the line that plays the
'pls-enter-vm-password' file.
>
> Steve
Why do that ? when you can simply replace the prompt file. Using your
method will need a recompile every time a different prompt needs to be
used.
>
>
> From: [EMAIL PROTECTED] on behalf of Java Rockx
> Sent: Mon 8/30/2004 8:10 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Newbie - Voicemail Password Help
>
>
>
> Hello All.
>
> I'm just beginning with Asterisk and I have it all working now. I'm
using
> Asterisk 1.0 RC1.
>
> My only question is this; when I check my voice mail the PBX simply
says
> "password". I wanted to make it say "please enter your voice mail
password" so
> I am using Background(pls-enter-vm-password).
>
> However now I hear "Please enter your voice mail password password"
when I
> check my messages.
>
> That's not a type-o. It says "password" twice.
>
> Here is my extensions.conf file.
>
> [macro-vmanswer]
>
>
> exten => s,1,GotoIf($[${ARG1} = ${CALLERIDNUM}]?2:5)
> exten => s,2,Background(pls-enter-vm-password)
> exten => s,3,VoicemailMain(${ARG1})
> exten => s,4,Hangup
> exten => s,5,Voicemail(u${ARG1})
> exten => s,6,Hangup
>
> [default]
> exten => 1002,1,Macro(vmanswer,1002)
>
>
>
>
> The whole point of the vmanswer macro is to go to the voice mail main
menu
> automatically when calling from your own phone, otherwise it sends
callers to
> the voice mail system to leave a message. Perhaps there's a better way
to do
> this as well. If so, please let me know.
>
> Regards,
> Paul
>
>
>
> __
> Do you Yahoo!?
> Yahoo! Mail Address AutoComplete - You start. We finish.
> http://promotions.yahoo.com/new_mail
> ___
> 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 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 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 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] Asterisk SIP between two networks
Hi all, I have next configuration: SIP Provider<--->ADSL router<---localnet 192.168.20.0--->ASTERISK<---localnet 172.24.240.0--->softphones first localnet 192.168.20.0 second localnet 172.28.240.0 in second localnet we have softphone and the first localnet is connected to ADSL router to connect to our SIP provider. The problem is the next: if I put in [general] section in sip.conf, bindaddr=0.0.0.0, I can't register in my SIP provider. If I put 192.168.20.10 in bindaddr I can register in my SIP provider but softphones can't register into asterisk. I 'm using asterisk RC1. Any idea? srsergio ___ 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] Zap & ANSWER the Call
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Nobody knows about that strange "behaviour" of Zap channels or
at least if is that right?
Thanks in advance.
Rodrigo P. Telles wrote:
| Hi,
|
| I'm using a TDM400 with one FXS and one FXO module (developer kit) and
| I've been testing termination from SIP phones to PSTN and it works fine,
| but
| asterisk accounting is doing something strange (for me).
| Scenario:
| 1 - extension 1009 (SIP phone - BT101)
| 2 - Zap/4-1 (TDM400 FXO module)
|
| extensions.conf:
| [dialout]
| exten => _9.,1,Dial(Zap/4/${EXTEN:1}|25|r)
| exten => _9.,2,Congestion
|
| [sip]
| include => dialout
| exten => 1009,1,Dial(SIP/1009,20,rt)
|
| So, when I dial "9something" from 1009, something rings and then I
| hangup the phone.
| I realised that asterisk thought that "something" ANSWERED the call:
|
| "","1009","9something","sip","""Tests""","SIP/1009-42fb","Zap/4-1","Dial",
| "Zap/4/something|25","2004-08-27 20:15:34","200
| 4-08-27 20:15:36","2004-08-27 20:15:43",9,7,"ANSWERED","BILLING"
|
| Is that right?
|
| Version: Asterisk 0.9.0
|
| Thanks in advance.
|
| --
|
| Rodrigo P. Telles <[EMAIL PROTECTED]>
| Project Manager
| Devel-IT - http://www.devel-it.com.br
| TDKOM Group
|
___
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
- --
Rodrigo P. Telles <[EMAIL PROTECTED]>
Project Manager
Devel-IT - http://www.devel-it.com.br
TDKOM Group
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.0.7 (GNU/Linux)
iD8DBQFBNPxViLK8unYgEMQRAnJ1AJ0WLPUHTMW9XJif5y5iECJJoloU8QCdHp1G
TGwaMVemCOPE1uOZQPFKdnc=
=zq37
-END PGP SIGNATURE-
___
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] Newbie - Voicemail Password Help
On Tuesday 31 August 2004 23:22, Umar Sear wrote: > On Tue, 2004-08-31 at 10:53, Stephen Hon wrote: > > Paul, > > > > What you can do is modify the source code for the voicemail application. > > > > Edit line 3579 in /usr/src/asterisk/apps/app_voicemail.c. Change the file > > 'vm-password' to 'pls-enter-vm-password'. > > > > Recompile and install. > > > > Then in your macro remove the line that plays the 'pls-enter-vm-password' > > file. > > > > Steve > > Why do that ? when you can simply replace the prompt file. Using your > method will need a recompile every time a different prompt needs to be > used. [...] It should be neither of those ways. The system should ideally be reading a definition file as to what files should be played and when. This could also allow for different languages. B Why can't people on this list delete the signatures? ___ 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] Newbie - Voicemail Password Help
On Tue, 2004-08-31 at 10:53, Stephen Hon wrote:
> Paul,
>
> What you can do is modify the source code for the voicemail application.
>
> Edit line 3579 in /usr/src/asterisk/apps/app_voicemail.c. Change the file
> 'vm-password' to 'pls-enter-vm-password'.
>
> Recompile and install.
>
> Then in your macro remove the line that plays the 'pls-enter-vm-password' file.
>
> Steve
Why do that ? when you can simply replace the prompt file. Using your
method will need a recompile every time a different prompt needs to be
used.
>
>
> From: [EMAIL PROTECTED] on behalf of Java Rockx
> Sent: Mon 8/30/2004 8:10 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Newbie - Voicemail Password Help
>
>
>
> Hello All.
>
> I'm just beginning with Asterisk and I have it all working now. I'm using
> Asterisk 1.0 RC1.
>
> My only question is this; when I check my voice mail the PBX simply says
> "password". I wanted to make it say "please enter your voice mail password" so
> I am using Background(pls-enter-vm-password).
>
> However now I hear "Please enter your voice mail password password" when I
> check my messages.
>
> That's not a type-o. It says "password" twice.
>
> Here is my extensions.conf file.
>
> [macro-vmanswer]
>
>
> exten => s,1,GotoIf($[${ARG1} = ${CALLERIDNUM}]?2:5)
> exten => s,2,Background(pls-enter-vm-password)
> exten => s,3,VoicemailMain(${ARG1})
> exten => s,4,Hangup
> exten => s,5,Voicemail(u${ARG1})
> exten => s,6,Hangup
>
> [default]
> exten => 1002,1,Macro(vmanswer,1002)
>
>
>
>
> The whole point of the vmanswer macro is to go to the voice mail main menu
> automatically when calling from your own phone, otherwise it sends callers to
> the voice mail system to leave a message. Perhaps there's a better way to do
> this as well. If so, please let me know.
>
> Regards,
> Paul
>
>
>
> __
> Do you Yahoo!?
> Yahoo! Mail Address AutoComplete - You start. We finish.
> http://promotions.yahoo.com/new_mail
> ___
> 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 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 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] Why is it called 'Comedian Mail?
I had always thought it was because an early clone of 'meridian mail' was called 'chameleon mail' and 'comedian mail' is a really good take off on 'chameleon mail'. ___ 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] Can asterisk detect BUSY signal?
Lol reverse hold! I can't see that working ever though, I tried it once and the agent at the other end hung up on me... I had to wait another hour in the queue... - Original Message - From: "Andrew Kohlsmith" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, August 31, 2004 2:53 PM Subject: Re: [Asterisk-Users] Can asterisk detect BUSY signal? > On Tuesday 31 August 2004 17:36, Kevin Walsh wrote: > > Spam-dialling should be made illegal. I, for one, wouldn't spend two > > seconds adding features to support this sort of usage. > > I can think of at least one legitimate use for this -- reverse spam dialling, > or at least "real person" detection. I hate sitting in hold queues and my > usual method is to put the phone on speaker and listen to Muzak while I wait. > > -A. > ___ > 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 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] Can asterisk detect BUSY signal?
On Tuesday 31 August 2004 17:36, Kevin Walsh wrote: > Spam-dialling should be made illegal. I, for one, wouldn't spend two > seconds adding features to support this sort of usage. I can think of at least one legitimate use for this -- reverse spam dialling, or at least "real person" detection. I hate sitting in hold queues and my usual method is to put the phone on speaker and listen to Muzak while I wait. -A. ___ 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] Why is it called 'Comedian Mail?
Chris Shaw [EMAIL PROTECTED] lazily top-posted: > I've wondered that myself... obviously the writer has a sense of humor! :) > > I like the sound of "Digium Mail", it sounds cool... > I like the sound of, err, nothing. Mine just prompts for "Mailbox?" -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] IAX Client
On Tue, 31 Aug 2004 14:14:40 -0400, "Jon Bebeau" <[EMAIL PROTECTED]> wrote: >Hello all, > >I'm working an a switchboard console for Asterisk and would like to investigate using >IAX Client library to Asterisk. I don't seem to be able to find the source. I'm >planning on a Win32 app. Guidance on where the source is or who to "take" to is >requested. > >Jon iaxclient.sf.net ___ 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] Why is it called 'Comedian Mail?
hmm Meridian Voice Mail == Comedian Voice Mail:) ___ 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] Can asterisk detect BUSY signal?
> suppose I have agents waiting on a queue and I configure asterisk to dial > out and to forward the call to the first agent enqueued. Asterisk will do > it even if the answer to the call is "busy". > > Is it possible to configure asterisk to detect the busy signal and, in > that case, dial another number, without wasting agent's time? > Spam-dialling should be made illegal. I, for one, wouldn't spend two seconds adding features to support this sort of usage. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] Why is it called 'Comedian Mail?
I've wondered that myself... obviously the writer has a sense of humor! :) I like the sound of "Digium Mail", it sounds cool... - Original Message - From: "Kevin Walsh" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Tuesday, August 31, 2004 2:30 PM Subject: RE: [Asterisk-Users] Why is it called 'Comedian Mail? > Kris Boutilier [EMAIL PROTECTED] wrote: > > Inquiring (management) minds want to know. I'm assuming it's because 'it's > > funny how simple it really is to write a really decent voicemail system'? > > > Perhaps it was written by someone with a red nose, oversized shoes and > a custard pie. I don't know either. > > -- >_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ > _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h > _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] > _/ _/ _/_/_/_/ _/_/_/_/ _/_/ > > ___ > 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 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] Why is it called 'Comedian Mail?
Kris Boutilier [EMAIL PROTECTED] wrote: > Inquiring (management) minds want to know. I'm assuming it's because 'it's > funny how simple it really is to write a really decent voicemail system'? > Perhaps it was written by someone with a red nose, oversized shoes and a custard pie. I don't know either. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ 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] T100P No D-channels
On Tue, 2004-08-31 at 15:58, Pliva, Josef wrote: > Unfortunately, I am seeing great many missed IRQs continually...if in fact > it is that which causes the loss of D-channel. Then you need to find out why interrupts are being locked for long enough to make the T100P miss interrupts. Common causes: frame buffer, graphics, RAID drivers, IDE DMA, and many, many other things. Doing an hdparm -u1 /dev/hda might help -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss." ___ 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] OT: Headset for Cisco 7960?
On Tue, 31 Aug 2004, Benjamin Johnson wrote: > I found the same with lots of headsets and my 7940, but I've just > plugged the headset from my Norstar system into the *handset* port on my > and it works perfectly. It's not ideal but it'll do for now! Ah, yeah, didn't think of that - works fine. On Tue, 31 Aug 2004, Dan Austin wrote: > GN-Netcom has a nice little headset for about US $120. As to the > pin-out, I believe that the headset port uses pins 1&4 instead of 2&3. I'll have to try building an adapter - shouldn't be tough. Thanks all! | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ 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] T100P No D-channels
H I guess from a troubleshooting standpoint to try and pinpoint the problem what I would do is remove all cards from the system and then only replace the cards that are absolutely necessary like your SCSI card and your Video card and of course the T100P and then check /proc/interrupts to see if you're having any more MISses... Also are you getting interrupt ERRs as well? Is APIC enabled for your board? If it is, you'll see things like IO-APIC-edge or IO-APIC-level in your /proc/interrupts. If not, you'll see XT-PIC for all interrupts... If you can't get APIC turned on you might try upgrading your kernel, your motherboard/bios may be blacklisted in that particular kernel... You definitely DO NOT want to share interrupts on the T100P unless it's a low-interrupt device like a USB controller or your video card. You definitely don't want to share with say a NIC or a SCSI controller... If it is sharing with one of those, try shuffling the cards around in different slots and make sure that your T100P isn't in slot 5. Slot 5 is usually shared with Slot 1 if they're on the same bus... Of course you've probably already tried all of this but just in case you haven't... -Chris ___ 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] Why is it called 'Comedian Mail?
Inquiring (management) minds want to know. I'm assuming it's because 'it's funny how simple it really is to write a really decent voicemail system'? Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District ___ 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] Hardware suggestion
Hi, Can anyone recommend a BRI card which works fine with asterisk and which supports point-to-point mode? Software fax detection should also work. Price does not matter. :) Digium seems to sell only PRI cards, and the Beronet drivers for the quad BRI cards seem to be in an early stage of development (besides, fax detection seems not to be implemented). Thanks pm ___ 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] T100P No D-channels
Hi Chris, thanks for taking time to look this over. T100P/* is connected to the Mitel IP-PBX/CU and it to telco - so I think our setting is correct. BTW, I did try 0 (as well as 2) without success, just for "fun", before I came on a good explanation of the sync source in this forum. Unfortunately, I am seeing great many missed IRQs continually...if in fact it is that which causes the loss of D-channel. Regards Josef -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: August 31, 2004 4:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] T100P No D-channels > Zaptel.conf sets t100p to be the primary sync source for the only span, as > suggested by many Asterisk users. I'm trying to understand so please bear with me... The T100P is connected directly to the Mitel? Or to the Telco through a T1? What I mean is are calls coming into the Mitel from the telco and then from there going into * or are calls going into * first and then being fed into the Mitel? If your T100P is connected to the telco then the clocking source should be the telco as their clocks are going to be a LOT more accurate than your PC's interrupt timers... If your T100P is connected to the Mitel, then you've got it right... Just checking, I wasn't sure from your description... Occasional interrupt misses are pretty normal although in a perfect world with a good mobo they should not happen at all... If you're seeing multiple misses per second (e.g. everytime you do cat /proc/interrupts you see more) then there's a problem... -Chris ___ 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 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] OT: Headset for Cisco 7960?
I use a Plantronics Supra H51 plugged straight into the headset port, and it works great. B. J. -Original Message- From: Nate Carlson [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 15:05 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] OT: Headset for Cisco 7960? Sorry, I know it's OT, but does anyone know of a relatively inexpensive headset that is compatible with the Cisco 7960? I've tried the headset off Norstar phones, doesn't seem to work with or without the amp. | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ 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] T100P No D-channels
> Zaptel.conf sets t100p to be the primary sync source for the only span, as > suggested by many Asterisk users. I'm trying to understand so please bear with me... The T100P is connected directly to the Mitel? Or to the Telco through a T1? What I mean is are calls coming into the Mitel from the telco and then from there going into * or are calls going into * first and then being fed into the Mitel? If your T100P is connected to the telco then the clocking source should be the telco as their clocks are going to be a LOT more accurate than your PC's interrupt timers... If your T100P is connected to the Mitel, then you've got it right... Just checking, I wasn't sure from your description... Occasional interrupt misses are pretty normal although in a perfect world with a good mobo they should not happen at all... If you're seeing multiple misses per second (e.g. everytime you do cat /proc/interrupts you see more) then there's a problem... -Chris ___ 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] OT: Headset for Cisco 7960?
Started a Wiki page here: http://www.voip-info.org/wiki-Cisco+Phone+Headsets Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Edward Eastman To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, August 31, 2004 10:28 AM Subject: RE: [Asterisk-Users] OT: Headset for Cisco 7960? Cisco headset pinout is different from normal ones (grr)If it's just for you, (ie nothing too professional ;) you can snip the leadof an existing plantronics type headset and do some reordering - this willgive you the necessary info (sorry - can't remember exactly how I did it):http://www.mml.uni-hannover.de/einhorn/headset/index_e.htmlIf you're after something more professional then obviously one of theleads/adapters will be a better approach.HTHEd-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of Nate CarlsonSent: 31 August 2004 21:05To: [EMAIL PROTECTED]Subject: [Asterisk-Users] OT: Headset for Cisco 7960?Sorry, I know it's OT, but does anyone know of a relatively inexpensiveheadset that is compatible with the Cisco 7960?I've tried the headset off Norstar phones, doesn't seem to work with orwithout the amp.| nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com || depriving some poor village of its idiot since 1981 |___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] OT: Headset for Cisco 7960?
Sorry, I know it's OT, but does anyone know of a relatively inexpensive headset that is compatible with the Cisco 7960? I've tried the headset off Norstar phones, doesn't seem to work with or without the amp. Plantonics S10 at Office Depot works fine. ___ 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] OT: Headset for Cisco 7960?
GN-Netcom has a nice little headset for about US $120. As to the pin-out, I believe that the headset port uses pins 1&4 instead of 2&3. Dan -Original Message- From: Edward Eastman [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 1:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] OT: Headset for Cisco 7960? Cisco headset pinout is different from normal ones (grr) If it's just for you, (ie nothing too professional ;) you can snip the lead of an existing plantronics type headset and do some reordering - this will give you the necessary info (sorry - can't remember exactly how I did it): http://www.mml.uni-hannover.de/einhorn/headset/index_e.html If you're after something more professional then obviously one of the leads/adapters will be a better approach. HTH Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nate Carlson Sent: 31 August 2004 21:05 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] OT: Headset for Cisco 7960? Sorry, I know it's OT, but does anyone know of a relatively inexpensive headset that is compatible with the Cisco 7960? I've tried the headset off Norstar phones, doesn't seem to work with or without the amp. | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ 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 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 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] T100P No D-channels
Hi Last week I installed Asterisk (release1) with digium t100p single span T1 (wct1xxp) board on Dell GX270 pc configured for PRI. Asterisk/t100p is currently the only user of the t1 line. All worked well for about a half a day, PSTN to SIP phones to non-SIP IP phones etc. Alas, since then I consistently get multitudes of blue alarms on all b-channels followed by a loss of d-channel: Aug 31 16:33:49 WARNING[98316]: chan_zap.c:5286 handle_init_event: Detected alarm on channel 1: Blue Alarm Aug 31 16:33:49 WARNING[98316]: chan_zap.c:5286 handle_init_event: Detected alarm on channel 2: Blue Alarm ...etc, intermixed with Aug 31 16:33:49 NOTICE[90123]: chan_zap.c:6920 pri_dchannel: PRI got event: 4 on Primary D-channel of span 1 Aug 31 16:33:49 WARNING[90123]: chan_zap.c:1899 pri_find_dchan: No D-channels available! Using Primary on channel anyway 24! ...and back to reset Aug 31 16:33:54 NOTICE[98316]: chan_zap.c:5281 handle_init_event: Alarm cleared on channel 1... ... Aug 31 16:33:54 NOTICE[90123]: chan_zap.c:6920 pri_dchannel: PRI got event: 5 on Primary D-channel of span 1 ... I found a few hits on VoIP.org and asterisk user forums usually mentioning PCI/BIOS IRQ sharing/conflict, but although I certainly see IRQ misses in zttool as well as /proc/zaptel/1, I cannot "see" any conflicts - zttool shows blue alarm, recovery and increasing IRQ misses right after zaptel/wct1xxp modprobe and ztcfg. During this search-for-the-truth I disabled all legacy devices (IRQs) I dared, including USB, but to no avail. On Dell GX270, BIOS does not seem to present the option of PCI IRQ line sharing/selection - just a disable/enable option. Mitel 3300 CU (part of 3300 IP-PBX) is set as pri_CPE and * t100p is pri_NET, using esf framing and b8zs code. Wildcard T100P shows green light, our 3300 Mitel CU light on the port I use ranges from yellow (during event recovery) to green (cleared). The telco rep sees nothing wrong with the Mitel - but did reset it several times since this problem started to happen, just to appease me. Zaptel.conf sets t100p to be the primary sync source for the only span, as suggested by many Asterisk users. No changes to Asterisk/Zaptel code has been done since the initial build from the Rel1 FTP site. After spending several days searching on internet, I found a lot of discussion about Digium PRI support which was not totally encouraging. However I am certain it is something simple since I am totally new to Asterisk environment and suspect I am missing something somewhere :( I would welcome any suggestions you may have. Thanks in advance Regards Josef [EMAIL PROTECTED] proc]# cat interrupts CPU0 0:6640846 XT-PIC timer 1:196 XT-PIC keyboard 2: 0 XT-PIC cascade 8: 1 XT-PIC rtc 9: 43296 XT-PIC eth0 10: 12708545 XT-PIC t1xxp 12: 1422 XT-PIC PS/2 Mouse 14: 84025 XT-PIC ide0 15: 256596 XT-PIC ide1 NMI: 0 ERR: 1 [EMAIL PROTECTED] proc]# cat pci PCI devices found: Bus 0, device 0, function 0: Host bridge: PCI device 8086:2570 (Intel Corp.) (rev 2). Prefetchable 32 bit memory at 0xe800 [0xefff]. Bus 0, device 1, function 0: PCI bridge: PCI device 8086:2571 (Intel Corp.) (rev 2). Master Capable. Latency=64. Min Gnt=8. Bus 0, device 30, function 0: PCI bridge: Intel Corp. 82801BA/CA/DB PCI Bridge (rev 194). Master Capable. No bursts. Min Gnt=2. Bus 0, device 31, function 0: ISA bridge: PCI device 8086:24d0 (Intel Corp.) (rev 2). Bus 0, device 31, function 1: IDE interface: PCI device 8086:24db (Intel Corp.) (rev 2). IRQ 9. I/O at 0x1f0 [0x1f7]. I/O at 0x3f6 [0x3f6]. I/O at 0x170 [0x177]. I/O at 0x376 [0x376]. I/O at 0xffa0 [0xffaf]. Non-prefetchable 32 bit memory at 0xfebffc00 [0xfebf]. Bus 0, device 31, function 2: IDE interface: PCI device 8086:24d1 (Intel Corp.) (rev 2). IRQ 9. I/O at 0xfe00 [0xfe07]. I/O at 0xfe10 [0xfe13]. I/O at 0xfe20 [0xfe27]. I/O at 0xfe30 [0xfe33]. I/O at 0xfea0 [0xfeaf]. Bus 0, device 31, function 3: SMBus: PCI device 8086:24d3 (Intel Corp.) (rev 2). IRQ 5. I/O at 0xefe0 [0xefff]. Bus 1, device 0, function 0: VGA compatible controller: PCI device 10de:0181 (nVidia Corporation) (rev 162). IRQ 11. Master Capable. Latency=64. Min Gnt=5.Max Lat=1. Non-prefetchable 32 bit memory at 0xfd00 [0xfdff]. Prefetchable 32 bit memory at 0xf000 [0xf7ff]. Bus 2, device 10, function 0: Network controller: Tiger Jet Network Inc. Model 300 128k (rev 0). IRQ 10. Master
RE: [Asterisk-Users] OT: Headset for Cisco 7960?
Cisco headset pinout is different from normal ones (grr) If it's just for you, (ie nothing too professional ;) you can snip the lead of an existing plantronics type headset and do some reordering - this will give you the necessary info (sorry - can't remember exactly how I did it): http://www.mml.uni-hannover.de/einhorn/headset/index_e.html If you're after something more professional then obviously one of the leads/adapters will be a better approach. HTH Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nate Carlson Sent: 31 August 2004 21:05 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] OT: Headset for Cisco 7960? Sorry, I know it's OT, but does anyone know of a relatively inexpensive headset that is compatible with the Cisco 7960? I've tried the headset off Norstar phones, doesn't seem to work with or without the amp. | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ 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 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] OT: Headset for Cisco 7960?
I found the same with lots of headsets and my 7940, but I've just plugged the headset from my Norstar system into the *handset* port on my and it works perfectly. It's not ideal but it'll do for now! Cheers, Benjamin Nate Carlson wrote: I've tried the headset off Norstar phones, doesn't seem to work with or without the amp. | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ 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 __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ 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] Streaming an audio file to a Zap channel before answer
Hi there
Background:
I want to add DDI and voicemail to users on an existing analogue pabx..
It does not support ISDN.
I have 10 DDI numbers via IAX which I am having sent to my Asterisk
box. I have 2 X100P cards connected to 2 analogue extension ports of my
main legacy analogue pabx. I have set up voicemail for each of my DDI
numbers, and when a call comes in for the person at pabx extension 21, I
do the following:
exten => 21,1,Macro(stdexten,21,Zap/g1c/21)
The c in the Dial command for Standard Extension causes the Zap channel
to not return answerbackto the calling party until the user presses a
'#' key to confirm answer. This is essential because in an
analogue-to-analogue call the only confirmation of answer is tones. I
don't want to use tone detection as it is too unrelaible and the UK
progress tones don't work well with callpogress detection anyway.
In my std-extension macro I include the Dial options r, to allow the
calling party to hear PSTN ringback until the channel is answered,
wither by the called party pressing # or the call going to voicemail.
exten => s,1,Dial(${ARG2},30,tTr); Ring the
interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1)
Everything works as expected, but there is one thing missing. The
called person picks up the phone and hears silence until they press the
# key to answer. This will really confuse my users. I therefore want
to play a helpful message _before the called person confirms answer,
along the lines of 'you have an incoming call. Press the hash key to
accept or hangup.' and loop this until either the person presses the #
key to accept the call, or the dial command times out and the call goes
to voicemail.
I have tried to work around this by using a Perl script in AGI, but the
AGI scripts seem to be single threaded, and "exec Background..." waits
til the background message has finished before moving on, defeating me.
Anyone got any ideas on this? Anyone hit a similar issue? Any
solutions out there?
Many thanks
Tim Robinson
Basingstoke UK
___
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] OT: Headset for Cisco 7960?
I don't know that Plantronics stuff qualifies as inexpensive but I have been using Plantronics H headsets with the adapter at this link. http://store.yahoo.com/founderstelecom/dirconcabfor.html I have two of these cables and they work very well. Bryan Nate Carlson wrote: Sorry, I know it's OT, but does anyone know of a relatively inexpensive headset that is compatible with the Cisco 7960? I've tried the headset off Norstar phones, doesn't seem to work with or without the amp. | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ 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 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] OT: Headset for Cisco 7960?
Sorry, I know it's OT, but does anyone know of a relatively inexpensive headset that is compatible with the Cisco 7960? I've tried the headset off Norstar phones, doesn't seem to work with or without the amp. | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ 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] Going to voicemail instead of queue if no agent is logged in ?
Hi, I'd like to implement scenario to send user to operator's queue by default (if doesn't dial any extension) but only if there is operator agent logged, so user could get response. If not I'd like to send it to voicemail... Any quick advice ? Thanks in advance, Robert. ___ 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] limit the length of extensions
On Wed, 2004-09-01 at 04:39, Deon Rodden wrote:
> All of my phones use sip, their accounts are in the sip.conf file and
> they have the context of 'company' or whatever. These phones need to be
> able to call each others extension, as well as dial outside to the real
> world. So in that context I put the outbound rules so that the phones
> can call out to the pstn, and I put the extensions of all the other
> phones in that context so that the phones can call each other.
>
> Different companies wanted it different. ie some wanted just local, or
> local and national, or local national and international. Some wanted to
> dial 9 to get an outside line, others wanted to be able to dial without
> the 9. So with the variance, I chose to put customized "outbound"
> extensions per context.You should *really* read the examples on the wiki and
> asterisk docs, and
other places, but, basically what you should do is this:
in sip.conf all your users belong to the context "inside-local" or "inside-ld" or
whatever.
[inside]
exten => 800,1,Dial...
exten => 801,1,Dial...
etc, or use a macro, or whatever you like
[remote]
include => inside
exten => s,1,PlayBack(menu)
etc
[dialout-local]
exten => _9XX,1,Dial(Zap/g1/${EXTEN:1})
exten => _XX,1,Dial(Zap/g1/${EXTEN})
[dialout-ld]
exten => _91NXXX,1,Dial(Zap/g1/${EXTEN:1})
exten => _1NXXX,1,Dial(Zap/g1/${EXTEN})
[inside-local]
include => inside
include => dialout-local
[inside-ld]
include => inside
include => dialout-local
include => dialout-ld
etc...
You should probably have another context for your internal applications
such as voicemail etc, another for global apps (maybe voicemail again,
or meetme, etc) and don't forget to include parking where appropriate.
Asterisk is powerful, and easy to make secure, however, like a lot of
other powerful devices, it is also easy to shoot yourself in the foot.
Regards,
Adam
___
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] Losing voice on Digium demo server - how to spot problem ?
Hi,
I'm trying to get Asterisk working on P4 2.8 server behind NAT and Firewall
(all ports we're set according to instructions) on DSL line.
When pbx connects to Digium demo server( I'm located in Slovenia, Europe),
it gets first few words, then silence and then comes back when enumerating
dial possibilities ("4 for accounting ...). Same happens from SIP or IAX
local extension.
I guess this is network problem, but would kindly ask for guidance for what
measures should I take and what seetings are first to try to avoid this
problems. I have another server running at my home on dialup line (28.8kbps)
and it connects to digium without problems, so I'm little suspicious being
only network traffic problem.
Thanks in advance for your effort,
regards,
Robert.
___
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] Can asterisk detect BUSY signal?
Nope, Asterisk will not do this, at least not without some serious busy-detect action going on and some tinkering with the dial and agents code, in which case any call that is not busy will have to wait a second or two for Asterisk to say that it isn't busy. Another way to go is to look into what the shady-dial people have done with agents/queues, they have probably already figured that part out. Or, you can try a different Asterisk-based predictive dialer: VICIDIAL, which is a part of the astGUIclient suite: http://astguiclient.sf.net/ It's got web-based management, a cross-platform GUI client, it'll run across multiple Asterisk servers and it's also free. MATT--- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 2:12 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Can asterisk detect BUSY signal? Hi, suppose I have agents waiting on a queue and I configure asterisk to dial out and to forward the call to the first agent enqueued. Asterisk will do it even if the answer to the call is "busy". Is it possible to configure asterisk to detect the busy signal and, in that case, dial another number, without wasting agent's time? Thanks ___ 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 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] Can asterisk detect BUSY signal?
[EMAIL PROTECTED] wrote: > Hi, > suppose I have agents waiting on a queue and I configure asterisk to > dial out > and to forward the call to the first agent enqueued. Asterisk will do > it even if > the answer to the call is "busy". > > Is it possible to configure asterisk to detect the busy signal and, > in that > case, dial another number, without wasting agent's time? Are you asking a "is this how it works" question, or have you tried using queue's and are not getting the intended results? It should be fairly easy to test, and determine what asterisk's response is. - Andrew Thompson http://aktzero.com/ ___ 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] detect telco voicemail stutter-tone
AFAIK, this is not possible - but I'll throw it out there anyhow... I subscribe to telco voicemail, for the event that all my pstn lines are in use. Telco gives me a stutter-tone dialtone when I have a message waiting. Can a Zap card detect this stutter-tone and perform some action? I'm using TDM400P+FXOs and SIP devices. Thanks ___ 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] limit the length of extensions
All of my phones use sip, their accounts are in the sip.conf file and
they have the context of 'company' or whatever. These phones need to be
able to call each others extension, as well as dial outside to the real
world. So in that context I put the outbound rules so that the phones
can call out to the pstn, and I put the extensions of all the other
phones in that context so that the phones can call each other.
Different companies wanted it different. ie some wanted just local, or
local and national, or local national and international. Some wanted to
dial 9 to get an outside line, others wanted to be able to dial without
the 9. So with the variance, I chose to put customized "outbound"
extensions per context.
Chris Shaw wrote:
Why are you including your outbound context into your incoming context in
the first place? That doesn't make any sense?
I'm guessing that because you're using a number in your exten => you're
using an IP channel like SIP or H323? Is this correct? If you're using a
T1/PRI or POTS lines you need to use 's'.
Using your example, your dialplan should look something like this...
[incoming]
exten => 9543340726,1,GotoIf($[${CALLERIDNAME} = anonymous]?2:4)
exten => 9543340726,2,setcidname(Blocked)
exten => 9543340726,3,setcidnum(00)
exten => 9543340726,4,Goto(companyname,beginmenu,1)
[companyname]
exten => beginmenu,1,SetVar(CALLEDNAME=CompanyName)
exten => beginmenu,2,Wait,1
exten => beginmenu,3,Answer() ; Answer the channel!
exten => beginmenu,4,Background(company-main)
exten => beginmenu,5,Background(ifyouknow)
exten => beginmenu,6,Goto(company_mainmenu,s,1)
exten => 502,1,Dial(SIP/whoever1&SIP/whoever2&sip/whoever3,30,m)
exten => 507,1,Dial(SIP/dave&SIP/jim&SIP/lisa,30,m)
...
[company_mainmenu]
exten => s,1,Background(company-nav1)
exten => 1,1,Goto(company_sales,s,1) ; Sales
exten => 2,1,Goto(companyname,502,1) ; Accounting
exten => 3,1,Goto(companyname,508,1) ; Customer Care
exten => 4,1,Goto(companyname,507,1) ; Technical Support
exten => 5,1,Goto(companyname,202,1) ; Human Resources
exten => 6,1,Goto(companyname,202,1) ; Provisioning
exten => 7,1,Goto(companyname,214,1) ; Marketing
exten => 0,1,Goto(companyname,210,1) ; Operator
...
Instead of jumping back and forth like this, I'd use macros to try and
condense the dialplan a bit...
I can help you more with this if you'd like...
Then for people inside the company there's this...
[outbound-local]
exten => _9NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten => _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,t) ; for
7-digit dialing
exten => _91800NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten => _91888NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten => _91877NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten => _91866NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
[outbound-ld]
exten => _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
[outbound-international]
exten => _9011.,Dial(SIP${EXTEN:[EMAIL PROTECTED],60,T)
[office]
include = outbound-local
include = outbound-ld
include = outbound-international
exten => _[1-5]XX,1,Dial(SIP/${EXTEN},25,tT) ; This is assuming they're all
SIP, you can use $DIALSTATUS to continue checking ZAP,MGCP,ETC...
and so on...
-Chris
___
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 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] IAX Client
Hello all, I'm working an a switchboard console for Asterisk and would like to investigate using IAX Client library to Asterisk. I don't seem to be able to find the source. I'm planning on a Win32 app. Guidance on where the source is or who to "take" to is requested. Jon ___ 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] Can asterisk detect BUSY signal?
Hi, suppose I have agents waiting on a queue and I configure asterisk to dial out and to forward the call to the first agent enqueued. Asterisk will do it even if the answer to the call is "busy". Is it possible to configure asterisk to detect the busy signal and, in that case, dial another number, without wasting agent's time? Thanks ___ 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] multiple lines with SIP like MGCP?
> We have a Dlink DVG-1120M and were surprised that it was able to handle 2 > simultaneous conversations to 2 seperate phones using only 1 MAC address and > 1 IP address. > > So we asked ourselves..why can't other 1 MAC/1IP devices do this as well? > > I have a Grandstream 486 that has 1IP and 1MAC. But I don't see anywhere in > sip.conf to add a second line to a device. Is this possible? Can this only > be done with an MGCP device? I don't have a Granstream, but the Cisco and Snom does that. There are no standards that dictate an IP per line. ___ 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] error: CDR on channel '' has not started
Hi, I installed asterisk-addons and configured it so that the cdr is done on a mysql database. Everything was fine, until I originated outgoing calls using the manager API. The call itself is performed perfectly, but when I hangup, I get the following warning on asterisk CLI: Aug 31 14:29:23 WARNING[-308995152]: cdr.c:331 ast_cdr_end: CDR on channel '' has not started Aug 31 14:29:23 WARNING[-308995152]: cdr.c:331 ast_cdr_end: CDR on channel '' has not started Aug 31 14:29:23 WARNING[-308995152]: cdr.c:478 ast_cdr_post: CDR on channel '' lacks start Aug 31 14:29:23 WARNING[-308995152]: cdr.c:118 ast_cdr_free: CDR on channel '' lacks start And the cdr record about this call is this: | | | | | | | | |||| 1969-12-31 21:00:00 | 1093973363 | 0 | UNKNOWN |0 | the date/time is always 1969-12-31/21:00:00. What could be wrong? I appreciate any help... Thank you very much ___ 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] Analog lines and TDM card
Hi, sorry to bother you, but i need to connect 8 standard analog lines to 2 asterisk servers (one in Italy (4 lines) and one in USA (4 lines)) and after let this 2 systems to interact between them. I was thinking to use the TDM400 card equipped with 4 FXO modules on both sides. Is it correct to do this (use the TDM card to terminate analog lines) or i have to use 4 X100P PCI card in both servers? Thank you in advance. Bye, Marcello ___ 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] multiple lines with SIP like MGCP?
The HT486 is a single-line device with a PSTN pass-thru. The only multiline IADs I know of are the SIPURAs and the Cisco ATA-186... What you do is you create 2 contexts, 1 for each line of the device and you set the host name to the IP address (or host name if applicable) of the IAD. Set the username of each context to the line's respective extension in Asterisk. Then in the web setup for the IAD, there should be a place to put the username for each line as well as the password... I have not tried this but it should work, SIP is not IP/MAC based it's more like SMTP, it's user based... -Chris ___ 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] Snom Programmable button Mini Howto and ringstate patch
It's very possible that the Polycom IP600 will work with this. As it is just an implementation of a SIP standard for subscribing to the state of other extensions. As for the feature improvements you might see them from me, but not very likely. It is easier for me to train my customers to hit *8 (I will probably just program a pickup button for them) than it is for me to figure out what I have to do in code to accomplish a call pickup. The conference stuff already works satisfactorily. If a person is on the phone you see their button lit, if you hit the button it calls them. They hit ok to accept your call and their existing call goes on hold. If they wish to conference they can this hit their conference button to bridge the three of you together. This is purely a function of the phone. More complex conferences I will achieve with use of the conference application and the flash control panel. You might, however, see the call parking bounty fulfilled by me when I get the time. David Hinkle -Original Message- From: John Todd [mailto:[EMAIL PROTECTED] Sent: Monday, August 30, 2004 4:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Snom Programmable button Mini Howto and ringstate patch At 1:23 PM -0500 on 8/30/04, David Hinkle wrote: >The snom 200 and 220 have five programmable buttons. Each button has a >led that can be used to indecate if an extension is idle, in use, or >ringing. A button pannel for the 220 is also comming out soon that will >have 20'ish programmable buttons on board. > >This is a killer app for any company that has receptionists handle >calls, and pretty usefull for everyone else. > >As a matter of fact, Asterisk already supports phone idle/in use states >for the buttons, and at the bottom of this message you will find a patch >to enable the ring state. > >Howto: > >1. Configure the programable buttons as "destination" and enter the >extension in the field. After saving the page the phone will convert >the extension to a sip url, which is fine. > >2. Modify your asterisk dialplan to provide "hints" that map a given >extension to a channel. (In asterisk, a channel can be busy or ringing, >but an extension is just a string of numbers that activate one or more >applications). Asterisk seems to provide syntax for allowing more than >one channel to be mapped to any particular extension with the hint >system, but I did not investigate that. > >Example: > >exten => 200,hint,SIP/RonC >exten => 200,1,Macro(stdexten,SIP/RonC) > >exten => 201,hint,SIP/JeanK >exten => 201,1,Macro(stdexten,SIP/JeanK) > >exten => 202,hint,SIP/JeffT >exten => 202,1,Macro(stdexten,SIP/JeffT) > >3. You must reload the dialplan and then reboot the phone for it's >subscriptions to take effect. After that, you should have working >lights. > >4. If you want the lights to blink on ringing, apply the following >patch to the asterisk code. > >You can not pick up a call by hitting the blinking button, I was going >to do this work but I decided to just train the receptionists to hit *8 >instead. I have not studied this extensivly, but to implement it, i >think it would just require asterisk to have support for sip "replaces" >(I don't know if asterisk supports this or not) and the ringing notify >needs to go out with a few more fields. (It seems that the snom phone >contacts the sip device listed in one of the ring notify message fields >with an invite including a "replaces" header to pick up a call) > >I have also included a sip trace of a snom phone picking up a call >placed to another phone using the blinking button in case anybody out >there wants to tackle this problem themselves (Sample trace was >collected when using snom phones with snom's sip proxy software). >Please note that it seems like we must include the extra fields in the >ring notify before the snom phone will procude the proper "replaces" >invite in order to do a standards compliant call pickup. > >Notes on patch: >If this patch is not in the proper format for submissions please provide >me a link to the asterisk submission policies. It has been tested here >at DerbyTech for about a week on our live phone system. > >I submit this patch to the asterisk project under the GPL with hope that >it will be resubmited to CVS. > >Thankyou, >David Hinkle >Sr. Linux Engineer >DerbyTech > This is pretty cool! I might get a Snom phone just to try them out. You asked for comments, so here are a few: 1) Send the patch in "diff -u" format; that's the format used in the bugtracker. 2) You'll need to sign the disclaimer on the http://bugs.digium.com/ interface. This disclaimer doesn't have much of a downside, and all patches to Asterisk for the public CVS have to be disclaimed in this way (avoids SCO-type lawsuits, etc.) 3) Have you looked at the configuration options for the Polycom IP600 phones? I don't know if this trick works with them, but they are pretty slick and have very
[Asterisk-Users] multiple lines with SIP like MGCP?
We have a Dlink DVG-1120M and were surprised that it was able to handle 2 simultaneous conversations to 2 seperate phones using only 1 MAC address and 1 IP address. So we asked ourselves..why can't other 1 MAC/1IP devices do this as well? I have a Grandstream 486 that has 1IP and 1MAC. But I don't see anywhere in sip.conf to add a second line to a device. Is this possible? Can this only be done with an MGCP device? Thanks, Matthew ___ 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] limit the length of extensions
Why are you including your outbound context into your incoming context in
the first place? That doesn't make any sense?
I'm guessing that because you're using a number in your exten => you're
using an IP channel like SIP or H323? Is this correct? If you're using a
T1/PRI or POTS lines you need to use 's'.
Using your example, your dialplan should look something like this...
[incoming]
exten => 9543340726,1,GotoIf($[${CALLERIDNAME} = anonymous]?2:4)
exten => 9543340726,2,setcidname(Blocked)
exten => 9543340726,3,setcidnum(00)
exten => 9543340726,4,Goto(companyname,beginmenu,1)
[companyname]
exten => beginmenu,1,SetVar(CALLEDNAME=CompanyName)
exten => beginmenu,2,Wait,1
exten => beginmenu,3,Answer() ; Answer the channel!
exten => beginmenu,4,Background(company-main)
exten => beginmenu,5,Background(ifyouknow)
exten => beginmenu,6,Goto(company_mainmenu,s,1)
exten => 502,1,Dial(SIP/whoever1&SIP/whoever2&sip/whoever3,30,m)
exten => 507,1,Dial(SIP/dave&SIP/jim&SIP/lisa,30,m)
...
[company_mainmenu]
exten => s,1,Background(company-nav1)
exten => 1,1,Goto(company_sales,s,1) ; Sales
exten => 2,1,Goto(companyname,502,1) ; Accounting
exten => 3,1,Goto(companyname,508,1) ; Customer Care
exten => 4,1,Goto(companyname,507,1) ; Technical Support
exten => 5,1,Goto(companyname,202,1) ; Human Resources
exten => 6,1,Goto(companyname,202,1) ; Provisioning
exten => 7,1,Goto(companyname,214,1) ; Marketing
exten => 0,1,Goto(companyname,210,1) ; Operator
...
Instead of jumping back and forth like this, I'd use macros to try and
condense the dialplan a bit...
I can help you more with this if you'd like...
Then for people inside the company there's this...
[outbound-local]
exten => _9NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten => _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,t) ; for
7-digit dialing
exten => _91800NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten => _91888NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten => _91877NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
exten => _91866NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
[outbound-ld]
exten => _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,T)
[outbound-international]
exten => _9011.,Dial(SIP${EXTEN:[EMAIL PROTECTED],60,T)
[office]
include = outbound-local
include = outbound-ld
include = outbound-international
exten => _[1-5]XX,1,Dial(SIP/${EXTEN},25,tT) ; This is assuming they're all
SIP, you can use $DIALSTATUS to continue checking ZAP,MGCP,ETC...
and so on...
-Chris
___
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] limit the length of extensions
You should be able to do that, but of course always test, test, test to make
sure.
Lyle
- Original Message -
From: "Deon Rodden" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Tuesday, August 31, 2004 11:24 AM
Subject: Re: [Asterisk-Users] limit the length of extensions
> If I put my outbound rules in a different context, and then "include"
> them in my main context, callers who call in will be able to access the
> extensions in the main context, but not the "included" (ie the outbound
> extensions) extensions called from the outbound context?
>
> Lyle Giese wrote:
>
> >You limit them by context. You put your outbound dialing patterns in a
> >context that inbound callers cann't access.
> >
> >Lyle
> >
> >- Original Message -
> >From: "Deon Rodden" <[EMAIL PROTECTED]>
> >To: <[EMAIL PROTECTED]>
> >Sent: Tuesday, August 31, 2004 9:05 AM
> >Subject: [Asterisk-Users] limit the length of extensions
> >
> >
> >
> >
> >>How do I limit the length of an extension? In my test IVR/Automated
> >>Attendant (whatever it's called), at the beginning it plays "if you know
> >>your parties 3 digit extension, you may enter it now) and then it gives
> >>a list of options. If the caller puts the 3 digit extension, it goes
> >>through fine, if they press 1, or 2 it goes to the selected menu option,
> >>but if they dial 91235551212 it dials that phone number. Which of
> >>course, is a big security risk.
> >>
> >>Is there a way to limit the length of an extension for an incoming call?
> >>My only solution right now is to duplicate ever single extension (about
> >>50 of them) in a seperate context, one that does not have the _9.
> >>extension in it, and then make the call in menu have access to that
> >>context. However, if I put a limit in the entire context of 3 digits,
> >>then my coworkers who's phones are in that context can only dial each
> >>other, not 9 and an outside number. So it has to be an incoming limit or
> >>something.
> >>
> >>Another possibly creative solution would be to "SetGroup(outsidecaller)
> >>on the incoming line" and then just before my outbound extension put
> >>"SetGroup(outsidecaller) and then a CheckGroup(2)" or something like
> >>that. I'd have to put another "SetGroup" in the outbound extension
> >>because there's no way to specify the group with the checkgroup command,
> >>it gets it from the setgroup statement.
> >>
> >>Any help would be appreciated.
> >>
> >>Thanks,
> >>Deon
> >>
> >>
> >>
> >>[incoming]
> >>exten => 9543340726,1,GotoIf($[${CALLERIDNAME} = anonymous]?2:4)
> >>exten => 9543340726,2,setcidname(Blocked)
> >>exten => 9543340726,3,setcidnum(00)
> >>exten => 9543340726,4,Goto(companyname,beginmenu,1)
> >>
> >>[companyname] ; All the phones, including outbound extensions are in
> >>this context
> >>exten => beginmenu,1,SetVar(CALLEDNAME=CompanyName)
> >>exten => beginmenu,2,Wait,1
> >>exten => beginmenu,3,Background(company-main)
> >>exten => beginmenu,4,Background(ifyouknow)
> >>exten => beginmenu,5,Goto(company_mainmenu,s,1)
> >>exten =>
> >>_9NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1})
> >>exten => 502,1,Dial(SIP/whatever1&SIP/whatever2|30|m)
> >>...
> >>
> >>[company_mainmenu]
> >>exten => s,1,Background(company-nav1)
> >>exten => 1,1,Goto(company_sales,s,1) ; Sales
> >>exten => 2,1,Goto(companyname,502,1) ; Accounting
> >>exten => 3,1,Goto(companyname,508,1) ; Customer Care
> >>exten => 4,1,Goto(companyname,507,1) ; Technical Support
> >>exten => 5,1,Goto(companyname,202,1) ; Human Resources
> >>exten => 6,1,Goto(companyname,202,1) ; Provisioning
> >>exten => 7,1,Goto(companyname,214,1) ; Marketing
> >>exten => 0,1,Goto(companyname,210,1) ; Operator
> >>
> >>___
> >>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 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 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 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] Re: Newbie - Voicemail Password Help
Thank you!
I took your advise and replaced the original vm-password.gsm file. Worked like
a charm.
Thanks again,
Paul
--- Jason Kawakami <[EMAIL PROTECTED]> wrote:
>
> - Original Message -
> >
> > Hello All.
> >
> > I'm just beginning with Asterisk and I have it all working now. I'm using
> > Asterisk 1.0 RC1.
> >
> > My only question is this; when I check my voice mail the PBX simply says
> > "password". I wanted to make it say "please enter your voice mail
> password" so
> > I am using Background(pls-enter-vm-password).
> >
> > However now I hear "Please enter your voice mail password password" when I
> > check my messages.
> >
> > That's not a type-o. It says "password" twice.
> >
> > Here is my extensions.conf file.
> >
> > [macro-vmanswer]
> >
> >
> > exten => s,1,GotoIf($[${ARG1} = ${CALLERIDNUM}]?2:5)
> > exten => s,2,Background(pls-enter-vm-password)
> > exten => s,3,VoicemailMain(${ARG1})
> > exten => s,4,Hangup
> > exten => s,5,Voicemail(u${ARG1})
> > exten => s,6,Hangup
>
> try
>
> exten => xxx,1,VoicemailMain(${CALLERIDNUM)
> exten => xxx,2,Hangup
>
> for your voicemailmain extension. this will recognize that your callerid if
> you have a mailbox on the system. note that your mailbox and your caller id
> must match
>
> alternatively, you could go into the /var/lib/asterisk/sounds directory and
> rename the file "vm-password" to "old.vm-password" then rename your file
> "pls-enter-vmpassword" to "vm-password"
>
> that way you would not have to alter the code at all.
>
>
> >
> > [default]
> > exten => 1002,1,Macro(vmanswer,1002)
> >
> >
> >
> >
> > The whole point of the vmanswer macro is to go to the voice mail main menu
> > automatically when calling from your own phone, otherwise it sends callers
> to
> > the voice mail system to leave a message. Perhaps there's a better way to
> do
> > this as well. If so, please let me know.
> >
> > Regards,
> > Paul
> >
>
> Good Luck
>
> Jason Kawakami
> www.optellabs.com
>
> ___
> 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
>
___
Do you Yahoo!?
Win 1 of 4,000 free domain names from Yahoo! Enter now.
http://promotions.yahoo.com/goldrush
___
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] Harddisk noise on TE410P
Claus: One difference is that I'm using the slower ATA disk, not the SCSI. Is the noise rhythmic (periodic) or constant? If periodic, what is the time between noise bursts? Do you hear the noise throughout a call, or just occasionally? Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California & London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Claus Futtrup Sent: Tuesday, August 31, 2004 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Harddisk noise on TE410P Hi there, The disks are SCSI Raid hotswap disks 1 RPM, P4 2.8 gig CPU, 1 Gig. of ram., and the server is running Red Hat 9.0. The sound is just like hearing a disk just muffled (sounds like strange static).. If you have a number I can call you at then you can hear it yourself. Kind Regards Claus Futtrup This message is for the designated recipient only and may contain privileged or confidential information. If you have received it in error, please notify the sender immediately and delete the original. Any other use of the email by you is prohibited. - Original Message - From: "Scott Stingel" <[EMAIL PROTECTED]> To: "'Claus Futtrup'" <[EMAIL PROTECTED]>; "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[EMAIL PROTECTED]> Sent: Tuesday, August 31, 2004 5:50 PM Subject: RE: [Asterisk-Users] Harddisk noise on TE410P > Claus- > > This is a problem that interests me, as I'm about to deploy TEN of these at > a customer site, all with TE410P's. > > I'm currently load testing one Proliant box (3GHz P4 processor) looping 59 > calls out to 59 calls in (leaving one channel open) - ie: lots of load. > While I'm doing this, I call in from another asterisk box over IAX, route > this call out over a TE410 channel and back in, and listen to a prompt. I > don't hear any unusual noise, and the box is performing well otherwise. > > Please supply more detail: What kind of disk, which Linux distro - and, what > is the noise you're hearing? > > Thanks > Scott Stingel > > > Scott M. Stingel > President, > Emerging Voice Technology, Inc. > Palo Alto California & London England > www.evtmedia.com > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Claus Futtrup > Sent: Tuesday, August 31, 2004 7:14 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Harddisk noise on TE410P > > Hi, > > I have this strange problem I need some help with.. It appears that I have > harddisk noise captured by a Digium TE410P card (Same problem on 2 identical > machines..) The machines are two Compaq Proliant DL320 G3's... > > Does anyone else have this problem.. > > Kind Regards > > Claus Futtrup > > > > --- > Outgoing mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.737 / Virus Database: 491 - Release Date: 11-08-2004 > > ___ > 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 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 > --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.737 / Virus Database: 491 - Release Date: 11-08-2004 ___ 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 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] PSTN noob question
> After reading a retarded amount of docs I'm still unable to figure out how to > get Asterisk to monitor my phone line and pick it up when the phone > rings...Im using a voice/fax/data modem on ttyS2. Any tips/pointers to > another stack of docs? Is this even doable without special hardware? No, its not possible with the modem you're talking about. Interfaces to a telephone line require an FXO interface, which can take the form of digium's hardware products (www.digium.com), external (ethernet attached) modules (1204 from www.mediatrix.com), ISDN adapters, etc. If you have a broadband internet connection, you can also sign up with several different providers that provide telephone numbers in many cities, extending those numbers to your asterisk box across your broadband internet service. Probably the least expensive what to play with asterisk is to purchase the x100p card from digium (supports one telephone line). You might dig around the http://www.voip-info.org/tiki-index.php (wiki) as there is a substantial amount of information on that site. Rich ___ 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] Harddisk noise on TE410P
Hi there, The disks are SCSI Raid hotswap disks 1 RPM, P4 2.8 gig CPU, 1 Gig. of ram., and the server is running Red Hat 9.0. The sound is just like hearing a disk just muffled (sounds like strange static).. If you have a number I can call you at then you can hear it yourself. Kind Regards Claus Futtrup This message is for the designated recipient only and may contain privileged or confidential information. If you have received it in error, please notify the sender immediately and delete the original. Any other use of the email by you is prohibited. - Original Message - From: "Scott Stingel" <[EMAIL PROTECTED]> To: "'Claus Futtrup'" <[EMAIL PROTECTED]>; "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[EMAIL PROTECTED]> Sent: Tuesday, August 31, 2004 5:50 PM Subject: RE: [Asterisk-Users] Harddisk noise on TE410P > Claus- > > This is a problem that interests me, as I'm about to deploy TEN of these at > a customer site, all with TE410P's. > > I'm currently load testing one Proliant box (3GHz P4 processor) looping 59 > calls out to 59 calls in (leaving one channel open) - ie: lots of load. > While I'm doing this, I call in from another asterisk box over IAX, route > this call out over a TE410 channel and back in, and listen to a prompt. I > don't hear any unusual noise, and the box is performing well otherwise. > > Please supply more detail: What kind of disk, which Linux distro - and, what > is the noise you're hearing? > > Thanks > Scott Stingel > > > Scott M. Stingel > President, > Emerging Voice Technology, Inc. > Palo Alto California & London England > www.evtmedia.com > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Claus Futtrup > Sent: Tuesday, August 31, 2004 7:14 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Harddisk noise on TE410P > > Hi, > > I have this strange problem I need some help with.. It appears that I have > harddisk noise captured by a Digium TE410P card (Same problem on 2 identical > machines..) The machines are two Compaq Proliant DL320 G3's... > > Does anyone else have this problem.. > > Kind Regards > > Claus Futtrup > > > > --- > Outgoing mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.737 / Virus Database: 491 - Release Date: 11-08-2004 > > ___ > 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 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 > --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.737 / Virus Database: 491 - Release Date: 11-08-2004 ___ 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] limit the length of extensions
If I put my outbound rules in a different context, and then "include"
them in my main context, callers who call in will be able to access the
extensions in the main context, but not the "included" (ie the outbound
extensions) extensions called from the outbound context?
Lyle Giese wrote:
You limit them by context. You put your outbound dialing patterns in a
context that inbound callers cann't access.
Lyle
- Original Message -
From: "Deon Rodden" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 31, 2004 9:05 AM
Subject: [Asterisk-Users] limit the length of extensions
How do I limit the length of an extension? In my test IVR/Automated
Attendant (whatever it's called), at the beginning it plays "if you know
your parties 3 digit extension, you may enter it now) and then it gives
a list of options. If the caller puts the 3 digit extension, it goes
through fine, if they press 1, or 2 it goes to the selected menu option,
but if they dial 91235551212 it dials that phone number. Which of
course, is a big security risk.
Is there a way to limit the length of an extension for an incoming call?
My only solution right now is to duplicate ever single extension (about
50 of them) in a seperate context, one that does not have the _9.
extension in it, and then make the call in menu have access to that
context. However, if I put a limit in the entire context of 3 digits,
then my coworkers who's phones are in that context can only dial each
other, not 9 and an outside number. So it has to be an incoming limit or
something.
Another possibly creative solution would be to "SetGroup(outsidecaller)
on the incoming line" and then just before my outbound extension put
"SetGroup(outsidecaller) and then a CheckGroup(2)" or something like
that. I'd have to put another "SetGroup" in the outbound extension
because there's no way to specify the group with the checkgroup command,
it gets it from the setgroup statement.
Any help would be appreciated.
Thanks,
Deon
[incoming]
exten => 9543340726,1,GotoIf($[${CALLERIDNAME} = anonymous]?2:4)
exten => 9543340726,2,setcidname(Blocked)
exten => 9543340726,3,setcidnum(00)
exten => 9543340726,4,Goto(companyname,beginmenu,1)
[companyname] ; All the phones, including outbound extensions are in
this context
exten => beginmenu,1,SetVar(CALLEDNAME=CompanyName)
exten => beginmenu,2,Wait,1
exten => beginmenu,3,Background(company-main)
exten => beginmenu,4,Background(ifyouknow)
exten => beginmenu,5,Goto(company_mainmenu,s,1)
exten =>
_9NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1})
exten => 502,1,Dial(SIP/whatever1&SIP/whatever2|30|m)
...
[company_mainmenu]
exten => s,1,Background(company-nav1)
exten => 1,1,Goto(company_sales,s,1) ; Sales
exten => 2,1,Goto(companyname,502,1) ; Accounting
exten => 3,1,Goto(companyname,508,1) ; Customer Care
exten => 4,1,Goto(companyname,507,1) ; Technical Support
exten => 5,1,Goto(companyname,202,1) ; Human Resources
exten => 6,1,Goto(companyname,202,1) ; Provisioning
exten => 7,1,Goto(companyname,214,1) ; Marketing
exten => 0,1,Goto(companyname,210,1) ; Operator
___
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 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 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] VoicePulse Connect DTMF with IAX2
My DID is 303 as well. Marty Mastera wrote: I'm using RC2 and last weekend's changes from VoicePulse. Outbound calling and dtmf works fine. However, an inbound call to my DID cannot send dtmf digits to the IVR. Thoughts? I have the same problem...my iax.conf is set up exactly as recommended per the recent Voicepulse changes and the configs they sent - my CVS is 7/14/04. Both inbound and outbound calling work, but no DTMF received on inbound calls. I found a post on the broadband reports forums regarding this issue, there where a few people who thought that it may affect VP customers who signed up for a DID in a new VP rate center/exchange...for example I've been waiting for VP to offer Colorado DID's (303 or 720) for quite awhile...so when I saw that they were available recently, I jumped on it and ordered one...so this a fairly new area code for them and I have the DTMF problem. I read other people that signed up for a fairly new area code having the same problem and emailing VP support to get it straightened out... I myself have sent them an email which they say they are checking into...I will be sure to let people know what my findings are. Marty ___ 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 -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.com ___ 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] PSTN noob question
After reading a retarded amount of docs I'm still unable to figure out how to get Asterisk to monitor my phone line and pick it up when the phone rings...Im using a voice/fax/data modem on ttyS2. Any tips/pointers to another stack of docs? Is this even doable without special hardware? TIA, 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] limit the length of extensions
You limit them by context. You put your outbound dialing patterns in a
context that inbound callers cann't access.
Lyle
- Original Message -
From: "Deon Rodden" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 31, 2004 9:05 AM
Subject: [Asterisk-Users] limit the length of extensions
> How do I limit the length of an extension? In my test IVR/Automated
> Attendant (whatever it's called), at the beginning it plays "if you know
> your parties 3 digit extension, you may enter it now) and then it gives
> a list of options. If the caller puts the 3 digit extension, it goes
> through fine, if they press 1, or 2 it goes to the selected menu option,
> but if they dial 91235551212 it dials that phone number. Which of
> course, is a big security risk.
>
> Is there a way to limit the length of an extension for an incoming call?
> My only solution right now is to duplicate ever single extension (about
> 50 of them) in a seperate context, one that does not have the _9.
> extension in it, and then make the call in menu have access to that
> context. However, if I put a limit in the entire context of 3 digits,
> then my coworkers who's phones are in that context can only dial each
> other, not 9 and an outside number. So it has to be an incoming limit or
> something.
>
> Another possibly creative solution would be to "SetGroup(outsidecaller)
> on the incoming line" and then just before my outbound extension put
> "SetGroup(outsidecaller) and then a CheckGroup(2)" or something like
> that. I'd have to put another "SetGroup" in the outbound extension
> because there's no way to specify the group with the checkgroup command,
> it gets it from the setgroup statement.
>
> Any help would be appreciated.
>
> Thanks,
> Deon
>
>
>
> [incoming]
> exten => 9543340726,1,GotoIf($[${CALLERIDNAME} = anonymous]?2:4)
> exten => 9543340726,2,setcidname(Blocked)
> exten => 9543340726,3,setcidnum(00)
> exten => 9543340726,4,Goto(companyname,beginmenu,1)
>
> [companyname] ; All the phones, including outbound extensions are in
> this context
> exten => beginmenu,1,SetVar(CALLEDNAME=CompanyName)
> exten => beginmenu,2,Wait,1
> exten => beginmenu,3,Background(company-main)
> exten => beginmenu,4,Background(ifyouknow)
> exten => beginmenu,5,Goto(company_mainmenu,s,1)
> exten =>
> _9NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1})
> exten => 502,1,Dial(SIP/whatever1&SIP/whatever2|30|m)
> ...
>
> [company_mainmenu]
> exten => s,1,Background(company-nav1)
> exten => 1,1,Goto(company_sales,s,1) ; Sales
> exten => 2,1,Goto(companyname,502,1) ; Accounting
> exten => 3,1,Goto(companyname,508,1) ; Customer Care
> exten => 4,1,Goto(companyname,507,1) ; Technical Support
> exten => 5,1,Goto(companyname,202,1) ; Human Resources
> exten => 6,1,Goto(companyname,202,1) ; Provisioning
> exten => 7,1,Goto(companyname,214,1) ; Marketing
> exten => 0,1,Goto(companyname,210,1) ; Operator
>
> ___
> 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 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] VoicePulse Connect DTMF with IAX2
On Tue, Aug 31, 2004 at 10:15:02AM -0400, Deon Rodden wrote:
> exten => _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
> exten => _1NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
> exten => _1NXXNXX,3,Congestion
This would dial the number twice..?
My config is
exten => _9.,1,Dial,IAX2/voicepulse/011${EXTEN:1}
exten => _9.,2,GotoIf($[ ${DIALSTATUS} != CONGESTION & ${DIALSTATUS} != CHANUNAVAIL
]?6)
exten => _9.,3,Dial,IAX2/voicepulse2/011${EXTEN:1}
exten => _9.,4,GotoIf($[ ${DIALSTATUS} != CONGESTION & ${DIALSTATUS} != CHANUNAVAIL
]?6)
exten => _9.,5,Congestion
exten => _9.,6,Hangup
arkadi.
___
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] PLC (Packet loss cancel) questions
> I have been reading the RFCs and I'm a bit more familiar with how it works > now although the algorithms are a bit over my head. I am somewhat new to > RTP/VoIP, but I have a strong telecom/networking background so it makes > things a bit easier to understand since they share a lot of common > features.. I just thought from the post mentioning only IAX2 and "some of > the other codecs" that SIP et. al. would be ignored... OOPS I meant... * protocols" that SIP et. al. would be ignored... ___ 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] VoicePulse Connect DTMF with IAX2
> I'm using RC2 and last weekend's changes from VoicePulse. Outbound > calling and dtmf works fine. However, an inbound call to my DID cannot > send dtmf digits to the IVR. > > Thoughts? I have the same problem...my iax.conf is set up exactly as recommended per the recent Voicepulse changes and the configs they sent - my CVS is 7/14/04. Both inbound and outbound calling work, but no DTMF received on inbound calls. I found a post on the broadband reports forums regarding this issue, there where a few people who thought that it may affect VP customers who signed up for a DID in a new VP rate center/exchange...for example I've been waiting for VP to offer Colorado DID's (303 or 720) for quite awhile...so when I saw that they were available recently, I jumped on it and ordered one...so this a fairly new area code for them and I have the DTMF problem. I read other people that signed up for a fairly new area code having the same problem and emailing VP support to get it straightened out... I myself have sent them an email which they say they are checking into...I will be sure to let people know what my findings are. Marty ___ 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] PLC (Packet loss cancel) questions
> This is nothing to do with SIP. It is an RTP issue, common to everything > which uses RTP - SIP and H.323 included. I have been reading the RFCs and I'm a bit more familiar with how it works now although the algorithms are a bit over my head. I am somewhat new to RTP/VoIP, but I have a strong telecom/networking background so it makes things a bit easier to understand since they share a lot of common features.. I just thought from the post mentioning only IAX2 and "some of the other codecs" that SIP et. al. would be ignored... >Sending no packets is perfectly valid, and normal, in RTP. If the receiving >end takes no packets (other than, perhaps, an extremely long silence) as a >disconnect it does not comply with the RTP spec. DTX is much despised, >and CNG only slightly better. They just sound good (pun intende) on paper. While I realize that hanging up on silence is not a desired behavior, unfortunately lots of things are out of spec... Look at Cisco's POE implementation for example, it's completely reversed from 802.3af specs... If * had at least some kind of continuous CNG capability it would help in these situations... Silence should be acceptable and even desired because it saves bandwidth, but apparently some people (and switches) find it uncomfortable... -Chris ___ 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] Harddisk noise on TE410P
Claus- This is a problem that interests me, as I'm about to deploy TEN of these at a customer site, all with TE410P's. I'm currently load testing one Proliant box (3GHz P4 processor) looping 59 calls out to 59 calls in (leaving one channel open) - ie: lots of load. While I'm doing this, I call in from another asterisk box over IAX, route this call out over a TE410 channel and back in, and listen to a prompt. I don't hear any unusual noise, and the box is performing well otherwise. Please supply more detail: What kind of disk, which Linux distro - and, what is the noise you're hearing? Thanks Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California & London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Claus Futtrup Sent: Tuesday, August 31, 2004 7:14 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Harddisk noise on TE410P Hi, I have this strange problem I need some help with.. It appears that I have harddisk noise captured by a Digium TE410P card (Same problem on 2 identical machines..) The machines are two Compaq Proliant DL320 G3's... Does anyone else have this problem.. Kind Regards Claus Futtrup --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.737 / Virus Database: 491 - Release Date: 11-08-2004 ___ 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 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] Polycom IP 300 - Displaying Only CallerNAME... What about NUMBER?
On Tue, 2004-08-31 at 10:37, Matthew Marlowe wrote:
> I'll do that, but I'm sure that it's the Polycom :) Caller ID shows up
> fine on my 7960... W/ the name on top and the number below that.
>
> -- Executing NoOp("SIP/614-3ede", "Caller*ID is Matthew Marlowe
> <6092521155>") in new stack
>
> When the phone rings, only 'Matthew Marlowe' would display. When I
> answer, both the Name & Number will show. It's simple while the phone
> is ringing that it doesn't display.
Is Matthew Marlowe in the Polycom directory application? Is so that
might be the reason it's not working as expected. I seem to recall
reading about it somwewhere in the Admin guide in the section about the
on phone directory/speed dial list
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."
___
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] Polycom IP 300 - Displaying Only CallerNAME... What about NUMBER?
I'll do that, but I'm sure that it's the Polycom :) Caller ID shows up
fine on my 7960... W/ the name on top and the number below that.
-- Executing NoOp("SIP/614-3ede", "Caller*ID is Matthew Marlowe
<6092521155>") in new stack
When the phone rings, only 'Matthew Marlowe' would display. When I
answer, both the Name & Number will show. It's simple while the phone
is ringing that it doesn't display.
I mean I doubt the polycom is malfunctioning, that's why I think there
might be some configuration to change... But what, I have no idea.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
Wieling
Sent: Tuesday, August 31, 2004 11:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom IP 300 - Displaying Only
CallerNAME... What about NUMBER?
Put a NoOp(Caller*ID is ${CALLERID}) in your dialplan JUST before the
Dial to the Polycom. See if the correct name and number shows up on the
console when the NoOp runs. If it does, there's a problem in the
Polycom, if there is no NAME then you have a problem with your Asterisk
config.
On Tue, 2004-08-31 at 10:08, Matthew Marlowe wrote:
> I got most of the features of my phone working. Polycom TEch support
> refuses to help or even talk to me. So I'll have to ask here again.
>
> On incoming calls, only the NAME is displayed. I am trying to figure
> out how to get the NAME & NUMBER displayed.
>
> If anyone can help me do this it would be GREATLY appreciated.
>
> Thank you in advance
> ___
> 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
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a
related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."
___
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 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] answer from wrong port
Hi everyone, I'm having a little problem and was wondering whether anyone would have any ideas or pointers for me. I've been working on load-balancing asterisk and have had a pretty successful setup using LVS and IP tunneling (plus a bit of iptables nating). I am only load balancing the SIP registration while the RTP between the SIP phone and the asterisk server and between the asterisk server and the CISCO AS5300 is being done directly with the real IP. Now this setup worked wonderfully, and I had tested with SIP phones behind different routers to see if Natting wasn't causing a problem and everything worked fine. But one of my locations recently changed routers (Linksys WRT54G) and the SIP phone no longer registers with the asterisk servers. After a bit of sniffing adn testing, here's what I came up with. If the phone connects directly to the asterisk server without load-balancing, it works fine. If the phone connects to the asterisk server through the load-balancing, the REGISTER packet comes into the asterisk server, but the reply instead of being sent-out from source-port 5060, it's sent out from source-port 1343 (or other lowest free port (1024,1026) and is blocked at the linksys gateway. Any ideas why asterisk doesn't use the 5060 source port in the reply? I'm unfortunately using version 0.9.0 of asterisk (my boss doesn't want to go with CVS). P.S. The iptables part of the load-balancing NATs the source IP of the reply packets as being from the virtual IP because asterisk sets it as from the real IP. The rest is "normal" lvs Thanks for any help Benjamin -- \\\|/// \\ - - // ( @ @ ) ---oOOo-(_)-oOOo--- There are times when truth is stranger than fiction and lunch time is one of them. --Oooo- oooO( ) Benjamin Benthos Lawetz ( ) ) /mailto:[EMAIL PROTECTED] \ ((_/ ICQ# 4269530 \_) ___ 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] Can i send calling costs to a SIP IP phone display
Is there a solution for asterisk to send the calling costs to a display of a grandstream Bt101 phone. Does anybody know if there is a solution for this? Greetings Han ___ 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] Polycom IP 300 - Displaying Only Caller NAME... What about NUMBER?
Put a NoOp(Caller*ID is ${CALLERID}) in your dialplan JUST before the
Dial to the Polycom. See if the correct name and number shows up on the
console when the NoOp runs. If it does, there's a problem in the
Polycom, if there is no NAME then you have a problem with your Asterisk
config.
On Tue, 2004-08-31 at 10:08, Matthew Marlowe wrote:
> I got most of the features of my phone working. Polycom TEch support
> refuses to help or even talk to me. So I'll have to ask here again.
>
> On incoming calls, only the NAME is displayed. I am trying to figure
> out how to get the NAME & NUMBER displayed.
>
> If anyone can help me do this it would be GREATLY appreciated.
>
> Thank you in advance
> ___
> 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
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."
___
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] Polycom SoundPoint... Gains -Which is for speakerphone
John, By chance do you know how to set a default ringer? What I have done is the following: As you can see, I want 7 to be the default ringer for line 1... For some reason, it doesn't take these changes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Baker Sent: Tuesday, August 31, 2004 11:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom SoundPoint... Gains -Which is for speakerphone 1) The samples are empty? No, they have variables with settings. Maybe I'm not understanding you. 2) I don't know how to dump the current settings to an xml file. You might try increasing the log level, but I doubt you're going to get a pretty looking xml file written to the log files. You're better off messing with the included config files. John Michael Graves wrote: > On Mon, 30 Aug 2004 22:09:26 -0500, John Baker wrote: > > >>Hmmm... >> >>Hands Free might be: >> >>voice.gain.rx.digital.chassis="15" (15 is my setting) >> >>Call waiting? You can turn it off in sip.cfg - do not disturb >>settings I think. Don't know about gain for call waiting. You might >>try playing with some of the variables in ipmid.cfg under >> >> >> >>John > > > Is there some way to get the phones current settings extracted to a > sample xml file? The sample files that come with the software > distributions are just empty frameworks with no settings. In my case I > have some IP600s. > > Thanks, > > Michael > -- > Michael Graves [EMAIL PROTECTED] > Sr. Product Specialist www.pixelpower.com > Pixel Power Inc. [EMAIL PROTECTED] > > o713-861-4005 > o800-905-6412 > c713-201-1262 > > It's a funny thing about life; if you refuse to accept anything but > the best, you quite often get it. ? W. Somerset Maugham > > ** Tag(s) inserted by Bandit Tagger98 - > http://www.gbar.dtu.dk/~c918704 > > > ___ > 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 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 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] Polycom SoundPoint... Gains - Whichis for speakerphone
If that was possible, that would make my life easier as well :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Tuesday, August 31, 2004 10:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom SoundPoint... Gains - Whichis for speakerphone On Mon, 30 Aug 2004 22:09:26 -0500, John Baker wrote: >Hmmm... > >Hands Free might be: > >voice.gain.rx.digital.chassis="15" (15 is my setting) > >Call waiting? You can turn it off in sip.cfg - do not disturb settings >I think. Don't know about gain for call waiting. You might try >playing with some of the variables in ipmid.cfg under > > > >John Is there some way to get the phones current settings extracted to a sample xml file? The sample files that come with the software distributions are just empty frameworks with no settings. In my case I have some IP600s. Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 It's a funny thing about life; if you refuse to accept anything but the best, you quite often get it. - W. Somerset Maugham ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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 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] Polycom IP 300 - Displaying Only Caller NAME... What about NUMBER?
I got most of the features of my phone working. Polycom TEch support refuses to help or even talk to me. So I'll have to ask here again. On incoming calls, only the NAME is displayed. I am trying to figure out how to get the NAME & NUMBER displayed. If anyone can help me do this it would be GREATLY appreciated. Thank you in advance ___ 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] Polycom SoundPoint... Gains - Which is for speakerphone
1) The samples are empty? No, they have variables with settings. Maybe I'm not understanding you. 2) I don't know how to dump the current settings to an xml file. You might try increasing the log level, but I doubt you're going to get a pretty looking xml file written to the log files. You're better off messing with the included config files. John Michael Graves wrote: On Mon, 30 Aug 2004 22:09:26 -0500, John Baker wrote: Hmmm... Hands Free might be: voice.gain.rx.digital.chassis="15" (15 is my setting) Call waiting? You can turn it off in sip.cfg - do not disturb settings I think. Don't know about gain for call waiting. You might try playing with some of the variables in ipmid.cfg under John Is there some way to get the phones current settings extracted to a sample xml file? The sample files that come with the software distributions are just empty frameworks with no settings. In my case I have some IP600s. Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 It's a funny thing about life; if you refuse to accept anything but the best, you quite often get it. ? W. Somerset Maugham ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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 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] which distro for asterisk?
On Tue, Aug 31, 2004 at 11:02:30AM +, Brian Wilkins wrote: > I had that problem, but apt-get install did the trick. Not to mention apt-get source and apt-get build-dep if you need to patch existing packages -- Tzafrir Cohen +---+ http://www.technion.ac.il/~tzafrir/ |vim is a mutt's best friend| mailto:[EMAIL PROTECTED] +---+ ___ 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] Polycom SoundPoint... Gains - Which is for speakerphone
On Mon, 30 Aug 2004 22:09:26 -0500, John Baker wrote: >Hmmm... > >Hands Free might be: > >voice.gain.rx.digital.chassis="15" (15 is my setting) > >Call waiting? You can turn it off in sip.cfg - do not disturb settings >I think. Don't know about gain for call waiting. You might try playing >with some of the variables in ipmid.cfg under > > > >John Is there some way to get the phones current settings extracted to a sample xml file? The sample files that come with the software distributions are just empty frameworks with no settings. In my case I have some IP600s. Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 It's a funny thing about life; if you refuse to accept anything but the best, you quite often get it. W. Somerset Maugham ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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] Re: Newbie - Voicemail Password Help
- Original Message -
>
> Hello All.
>
> I'm just beginning with Asterisk and I have it all working now. I'm using
> Asterisk 1.0 RC1.
>
> My only question is this; when I check my voice mail the PBX simply says
> "password". I wanted to make it say "please enter your voice mail
password" so
> I am using Background(pls-enter-vm-password).
>
> However now I hear "Please enter your voice mail password password" when I
> check my messages.
>
> That's not a type-o. It says "password" twice.
>
> Here is my extensions.conf file.
>
> [macro-vmanswer]
>
>
> exten => s,1,GotoIf($[${ARG1} = ${CALLERIDNUM}]?2:5)
> exten => s,2,Background(pls-enter-vm-password)
> exten => s,3,VoicemailMain(${ARG1})
> exten => s,4,Hangup
> exten => s,5,Voicemail(u${ARG1})
> exten => s,6,Hangup
try
exten => xxx,1,VoicemailMain(${CALLERIDNUM)
exten => xxx,2,Hangup
for your voicemailmain extension. this will recognize that your callerid if
you have a mailbox on the system. note that your mailbox and your caller id
must match
alternatively, you could go into the /var/lib/asterisk/sounds directory and
rename the file "vm-password" to "old.vm-password" then rename your file
"pls-enter-vmpassword" to "vm-password"
that way you would not have to alter the code at all.
>
> [default]
> exten => 1002,1,Macro(vmanswer,1002)
>
>
>
>
> The whole point of the vmanswer macro is to go to the voice mail main menu
> automatically when calling from your own phone, otherwise it sends callers
to
> the voice mail system to leave a message. Perhaps there's a better way to
do
> this as well. If so, please let me know.
>
> Regards,
> Paul
>
Good Luck
Jason Kawakami
www.optellabs.com
___
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
