Re: [Asterisk-Users] Meetme2 compilation problem
User=guest, password=restricted.This account wil be open util friday.Nope:220 Welcome to the Vink Consultancy FTP server. Please login...Name (ftp.vinkconsult.com:brianc): guest331 Password required for guest.Password:530 Login incorrect.Login failed.Yep: $ ftp ftp://guest:[EMAIL PROTECTED]/Connected to services.vinkconsult.com.220 Welcome to the Vink Consultancy FTP server. Please login...Remote system type is UNIX.Using binary mode to transfer files.331 Password required for guest.230 User guest logged in.200 Type set to I ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Call waiting in Australia
Has anyone had problems with Call Waiting signals causing Zap channel or bridging hangups in AU. I was on a call the other day (Zap channel to PSTN) and the call suddenly hung up on my side. I dialled the calling party and got the call again, it seems that the bridge had dropped and that the other party had not lost the connection. As soon as I got the bridging again the other party mentioned that they had had a call waiting signal immediately before I went off the air. Any one had similar experiences, or have fixes? I'm in Australia, I have the same setup, and I had the exact same thing happen twice in the space of a few minutes, just then, while calling the same person. The person who I was calling says they don't have call waiting and were disconnected from me without warning, as I was. I have disabled call waiting with my telco. I rebooted asterisk today. Personally, I've come to the conclusion that these digium cards are a bit flaky - dunno? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do I do this ?
I wish to initate calls from a web interface, by clicking on a link and then connecting to the automatic outgoing call by picking up an analogue phone. I've got one fxs and one fxo and I wish to automate the call using a call file (which I can do now). How can I pick up a handset and connect to this call I've made when it's ringing? Can someone point me as to how I may be able to do this. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Automating calls
Hi, I wish to initate calls from a web interface, by clicking on a link and then connecting to the automatic outgoing call by picking up an analogue phone. I've got one fxs and one fxo and I wish to automate the call using a call file (which I can do now). How can I pick up a handset and connect to this call I've made when it's ringing? Can someone point me as to how I may be able to do this. Thanks, Oliver ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Modem as PSTN interface?
Hi, all Can a normal PCI modem be used to provide PSTN interface? I have seen modems that have answering machine capabilities, so there should not be a problem sending voice through them. Certainly, modem will be cheaper option then dedicated cards. Am I missing something? Most modems don't operate at full-duplex. A normal modem can be used to send voice, then switch to recording voice, but it can't send and receive simultaneously. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Who makes these phones?
http://www.broadbandphone.com.au/global/pnp.htm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Caller ID in AU
Is anyone in AU successfully getting Caller ID from the analogue PSTN service? If so, what settings? -- Howard. http://www.voip-info.org/tiki-index.php?page=Asterisk+and+Australian+Caller+ID ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] International area codes (incl. mobile)
Hello everybody, does anybody knows from where I can get an list of international area codes incl. the mobile numbers? Have you tried google ? http://www.google.com.au/search?hl=enq=international+dialing+codes ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Extensions to solve three way calling problem
On Wed, 2005-01-05 at 18:38 +1100, PHP Mechanic wrote:
Have you considered setting up a meetme confrence line for them? :)
analog phone = asterisk/tdm11b = pstn
I have played with it. But the problem I'm having is as follows
exten = _1800.,1,Dial(Zap/4/${EXTEN},20,Tr) ; call some company
willing to pay for my test, preferably get someone with an on hold
message
; Now I press #* on the analog phone to transfer them to Meetme
exten = *,1,Meetme,2000 ; send
them to meetme
exten = *,2,Flash()
;
flash the pstn line
What makes you think that would flash the PSTN line?
Because the cli reports that it is executing flash on the Zap/4 - the PSTN
line
This is your
problem. When you transfer the PSTN line anywhere and then go to dial
again, the flash is actually on the current channel. I wouldn't be
surprised if you hear it in your receiver. I don't know of anyway to
flash the PSTN line from within asterisk that would do as you want. In
fact, to enable it would be a security risk as well. Think of the
possibility of having multiple lines in and then dialing an extension to
flash the line and messing up and flashing someone else's connection.
Closest thing I could think of is having your PSTN side caller do the
transfer and redial. If the PSTN caller was allowed to transfer the
inside person and then dial a special extension that would initiate the
flash and the dial command. Of course the trouble here is that as soon
as the flash occurs, the new caller is the one going to be stuck in an
odd state and the previous PSTN caller is going to be in unrecoverable
limbo.
Just looks like you will be SOL on utilizing the PSTN 3 way calling.
Yeah, I think you are right.
But what is the point of threewaycalling and transfer in zapata.conf - what
do they do?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
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] Extensions to solve three way calling problem
Threeway calling is similar. You can make a small impromptu conference that way with 2 internal phones and an external or 3 internal phones or even 1 internal and 2 external calls on separate phone lines. All of these are mixed inside of asterisk and the PSTN is non the wiser. Thanks for clearing this up for me. The thing that still get's me is that the pstn is a little bit wise. It can perform the following: 1. Establish a call with the first person. You can call them or they can call you. 2. Press Flash/Recall on phone to put the first person on hold. 3. Wait until you hear the dial tone. 4. Dial the number of the second person. 5. Wait until you hear the second line ringing. 6. Press Flash/Recall and talk to the first person (they will hear the ringing tone too). Can I make asterisk play ball with my telco? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] CallerID in Australia Analogue PSTN Phone
Is there anyone using * in AU that has successfully extracted the CLID
from an incoming analogue PSTN phone call, and would like to spread
the
word?
What I need more though is examples of anything that needs to go into
extensions.conf
You could add this line if you want
exten = s,1,NoOp(Caller ID on the PSTN line is ${CALLERID})
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
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] CallerID in Australia Analogue PSTN Phone
What I need more though is examples of anything that needs to go into
extensions.conf
You could add this line if you want
exten = s,1,NoOp(Caller ID on the PSTN line is ${CALLERID})
M. Tried that, but it didn't deliver ${CALLERID}
Did the caller have callerid enabled by their telco ?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extensions to solve three way calling problem
My daughter hates my phone system because she can't use the three way function. What should my extension.conf look like to solve this problem. Without asterisk I can do the following without any trouble. 1. Establish a call with the first person. You can call them or they can call you. 2. Press Three Way, or Flash/Recall on any other touch tone phone to put the first person on hold. 3. Wait until you hear the dial tone. 4. Dial the number of the second person. 5. Wait until you hear the second line ringing. 6. Press Three Way or Flash/Recall and talk to the first person (they will hear the ringing tone too). However I have the following setup and I want to be able to do the same thing and it just doesn't work. analog phone = asterisk/tdm11b = pstn Can this be done? Does anyone have some extensions they could share with me which could achieve the same thing. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Extensions to solve three way calling problem
From: Dr. Matthew Roller Have you considered setting up a meetme confrence line for them? :) I've tried that and I get as far as flashing the pstn after a transfer: ie: Hit the '#' and then '1' on the analogue phone. exten = 1,1,Flash() exten =1,2,MeetMe,2000 The problem is that after transfering the first caller to MeetMe I am unable to make an outgoing call using the pstn line, I don't get the dialtone and if I hang up and dial out I get a everyone is busy message, because I'm assuming that the line is still tied up with the first caller who is now in MeetMe. This problem has been bugging me for weeks. It's possible to do without using asterisk and all the settings in zapata.conf seem to suggest it's possible. But I am unable to make it work. Any other suggestions welcome. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] CallerID in Australia Analogue PSTN Phone
Howard Lowndes wrote: Is there anyone using * in AU that has successfully extracted the CLID from an incoming analogue PSTN phone call, and would like to spread the word? Yes, I have. I'm using a Voicetronix OpenLine4 card. I did have an issue with the config file, you have to set callerid=yes before each channel, unless you're running CVS from 2004/12/13 21:04:12 or later. What hardware are you using? chan_vpb has useful debugging info for callerid at debug level 4. I fixed it using this: http://www.voip-info.org/tiki-index.php?page=Asterisk+and+Australian+Caller+ID ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] CallerID in Australia Analogue PSTN Phone
I fixed it using this: http://www.voip-info.org/tiki-index.php?page=Asterisk+and+Australian+Caller+ID Actually settings such as these should really be moved to a settings header file. Please elaborate... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Extensions to solve three way calling problem
Have you considered setting up a meetme confrence line for them? :)
analog phone = asterisk/tdm11b = pstn
The meetme option is nice, but it doesn't solve the problem. The TDM11B
only
has one FXO, one FXS. To get the effect the daughter wants requires
supporting the threeway facility the telco offers. You need to Flash the
outside line. Zap does have an application for that, but I haven't played
with what it can do, or how to program it.
I have played with it. But the problem I'm having is as follows
exten = _1800.,1,Dial(Zap/4/${EXTEN},20,Tr) ; call some company
willing to pay for my test, preferably get someone with an on hold message
; Now I press #* on the analog phone to transfer them to Meetme
exten = *,1,Meetme,2000 ; send
them to meetme
exten = *,2,Flash() ;
flash the pstn line
At this point they are moved to meetme and the pstn line is flashed.
I am disconnected and don't get a dialtone, I get a busy tone.
So I physically hang up the phone and then attempt to call another 1800
number
When I do this the cli displays a notice:
Unable to create channel of type 'Zap' == Everyone is busy/congested at
this time
It is possible for me to join the meetme by dialing 2000 but I can't appear
to access the pstn line that I flashed
What I don't understand is that there are settings in zapata.conf for
threewaycalling, transfer, etc... but it doesn't seem to work and I wonder
what these settings do? But more importantly is there a way to solve my
problem?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hook/Flash, Hold, Call Waiting, Three Way Calling
Hi, I have a TDM411B and when I am using asterisk I can't get hook/flash or hold to work when using asterisk, which means I can't use three way calling or the call waiting functions. I've tried using combinations of hook flash button and *0 on three different phones and I dont get a dial tone, the other party is not put on hold, and I don't see the keys I'm pressing in the CLI. When I take asterisk out of the equasion and plug the analoge phones directly into the telephone line everything works as you would expect. Can someone post an example of a working extensions.conf / zapata.conf where they use hook/flash that I can try. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Hook/Flash, Hold, Call Waiting, Three Way Calling
For threeway calling (analog phone) I just hit the flash button get a dial tone, dial the number and hit the flash key again. It doesn't work for me when I'm using asterisk. No problems without it. So is my hardware broken or my dialplan? When you hit the flash key is anything displayed in the CLI ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Hook/Flash, Hold, Call Waiting, Three Way Calling
Hi, I have a TDM411B and when I am using asterisk I can't get hook/flash or hold to work when using asterisk, which means I can't use three way calling or the call waiting functions. I've tried using combinations of hook flash button and *0 on three different phones and I dont get a dial tone, the other party is not put on hold, and I don't see the keys I'm pressing in the CLI. When I take asterisk out of the equasion and plug the analoge phones directly into the telephone line everything works as you would expect. Can someone post an example of a working extensions.conf / zapata.conf where they use hook/flash that I can try. Do you have the following in your /etc/asterisk/zapata.conf BEFORE the channel number? Yes. My zapata.conf is below. My setup is POTS - Asterisk/TDM411B - PSTN Line. While connected to another party if I press flash or *0 on my analoge phone I don't get a dialtone, the called party does not go on hold and I don't see anything in the CLI - it doesn't work. It all works fine when I don't use asterisk. [channels] ; ; TDM400P Port #4 plugged wall ; This is the PSTN Line ; context=PSTN signalling=fxs_ls busydetect=yes ; to test when a line is hung-up busycount=6 ; to prevent suprious hangups echotraining=800 echocancel=yes immediate=no musiconhold=default usecallerid=yes callerid=asreceived channel = 4 ; ; TDM400P Port #1 plugged into analog Phone ; This phone is allowed to dial extensions and local and long distance numbers ; context=RealPhone signalling=fxo_ls callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no musiconhold=default usecallerid=yes callerid=Real Phone 1 mailbox=1 channel = 1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P callwaiting, threewaycalling and cancallforward problem
Hi, I have a TDM400P with one FXO and an FXS. My Australian telco has enabled call waiting and three way calling. When I'm using my telephone connected to the FXO I can see/hear on my handset that call waiting is enabled, however when I attempt to send a hook flash or use the combination *0 I don't get a dialtone and the feature doesn't work. The same goes for three way calling and cancallforward. I hope someone can explain to me what may be going on as it's been something I haven't been able to solve. My zapata conf is below. zapata.conf [channels] busydetect=yes ; to test when a line is hung-up busycount=6 ; to prevent suprious hangups echotraining=800 echocancel=yes immediate=no usecallerid=yes relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived signalling=fxs_ls group=1 context=PSTN channel = 4 signalling=fxo_ls mailbox=1 callerid=Real Phone 1 context=RealPhone channel = 1 ___ 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] TDM400P callwaiting, threewaycalling and cancallforward problem
When I'm using my telephone connected to the FXO I can see/hear on my handset that call waiting is enabled, however when I attempt to send a hook flash or use the combination *0 I don't get a dialtone and the feature doesn't work. The same goes for three way calling and cancallforward. Are you sure you are doing the right sequence of buttons. If you have a call coming in you must flash then * 0. the flash button flashes the PBX and * 0 flashes the CO. Yes just tried it again. Using the fxo I dialled a sip phone on the network, then I tried to send a flash and then *0 and it refuses to give me a dialtone. I just hear lots of beeping and I stay connected and don't get a dialtone. I also can't see anything showing up on the console. ___ 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 - RE: [Asterisk-Users] CallerID+Distinctive ring in Australia
This works for me: http://www.voip-info.org/tiki-index.php?page=Asterisk+and+Australian+Caller+ID Is this on analog or ISDN? A few of us discussed this a short while ago (call ID/Aust/analog) but nobody ever got anywhere. But I am sure that one of us will figure it all out soon enough. Later, PaulH Hawthorn Aust -Original Message- From: Elliot Mackenzie [mailto:[EMAIL PROTECTED] Sent: Tuesday, 9 November 2004 6:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CallerID+Distinctive ring in Australia Whoops. Apologies for sending this more than once. I thought a sendmail upgrade had broken, but it was just slow :-) Someone mentioned there are some patches required to make callerid work in the UK. Do similar patches apply for use in Australia? How would I make my own modifications? Is there some way I can record the analog signal incoming and play it back to figure out what asterisk is doing with it? Kind regards, Elliot. Elliot Mackenzie wrote: I have a situation involving both caller id and distinctive ring in australia that appears to be having issues. I am using the CVS snapshot current as of an hour ago. The distinctive ring was working ok until i arranged for the telco to turn on callerid: now the distinctive ring detection appears flaky and there is no callerid detected. By flaky distinctive ring, I mean the three numbers that are being detected on an incoming call (ever since callerid was switched on), vary from [0,0,0] to more sensible numbers (sensible, though they still vary by too much to make sense of). Once I have seen callerID returned with error, but this only happened once and said nothing further. I haven't replicated this message when I have had the debugging cranked right up unfortunately. There was a newspost a while back that suggested increasing a delay in callerid.c, which I have done but to no avail. I increased it from 4000 to 5600, as per the original post. Does anyone have ideas for what to try, or for how to start digging further? This problem has been plaguing me for a month now. Kind regards, Elliot. ___ 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 CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ 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 - Re: [Asterisk-Users] CallerID+Distinctive ring in Australia
Elliot Mackenzie wrote: I have a situation involving both caller id and distinctive ring in australia that appears to be having issues. I am using the CVS snapshot current as of an hour ago. The distinctive ring was working ok until i arranged for the telco to turn on callerid: now the distinctive ring detection appears flaky and there is no callerid detected. By flaky distinctive ring, I mean the three numbers that are being detected on an incoming call (ever since callerid was switched on), vary from [0,0,0] to more sensible numbers (sensible, though they still vary by too much to make sense of). Once I have seen callerID returned with error, but this only happened once and said nothing further. I haven't replicated this message when I have had the debugging cranked right up unfortunately. There was a newspost a while back that suggested increasing a delay in callerid.c, which I have done but to no avail. I increased it from 4000 to 5600, as per the original post. Does anyone have ideas for what to try, or for how to start digging further? This problem has been plaguing me for a month now. Have you tried this? http://www.voip-info.org/tiki-index.php?page=Asterisk+and+Australian+Caller+ID ___ 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] zapata.conf - callwaiting
Hi, I have callwaiting enabled by my telco, but my wife hates callwaiting so I tried to switch it off in * but it doesn't work: zapata.conf [channels] busydetect=yes ; to test when a line is hung-up busycount=6 ; to prevent suprious hangups echotraining=800 echocancel=yes immediate=no usecallerid=yes relaxdtmf=yes callwaiting=no callwaitingcallerid=no threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived signalling=fxs_ls group=1 context=PSTN channel = 4 signalling=fxo_ls mailbox=1 callerid=Real Phone 1 context=RealPhone channel = 1 How do I switch off callwaiting in asterisk? What am I missing? ___ 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] threeway calling not working
Hello, I'm having a few problems with getting zapatta.conf to work properly: threeway calling is enabled by my telco and works when not connected to asterisk. The problem is that threeway calling doesn't work nor does call forwarding. With threeway calling I am unable to send a hook/flash to asterisk and when I try call forwarding I can set it but the call does not get forwarded I get a everyone is busy message and then the call gets directed to voicemail. [channels] busydetect=yes ; to test when a line is hung-up busycount=6 ; to prevent suprious hangups echotraining=800 echocancel=yes immediate=no usecallerid=yes relaxdtmf=yes callwaiting=no callwaitingcallerid=no threewaycalling=yes transfer=yes cancallforward=yes callerid=asreceived signalling=fxs_ls group=1 context=PSTN channel = 4 signalling=fxo_ls mailbox=1 callerid=Real Phone 1 context=RealPhone channel = 1 ___ 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] Call Parking Problem
Hi,
I'm unable to pick up parked calls after they are transfered.
I get the transfer message when I press # and then I'm told 701 The
extension I'm dialing goes to the on hold music. I'm disconnected, I hang
up, dial 701 and I see this message on the console Everyone is
busy/congested at this time
I just have the default parkedcalls file, and have this in the extensions.
[AnalogPhone]
exten = _70X,1,Dial(Zap/1/${EXTEN},20,Ttr)
include = parkedcalls
[SipPhone]
exten = _70X,1,Dial(SIP/1/${EXTEN},20,Ttr)
include = parkedcalls
___
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 - RE: [Asterisk-Users] Call Parking Problem
That fixed it. Thanks
The 'parkedcalls' code dynamically creates and deletes entries in the
dialplan to handle the calls that have been parked, so the parking lot
must
not overlap your regular extensions. The initial parking extension is
statically created on startup, thus the 'exten =' entry is matching the
parking slot digits and throwing the congestion error - remove it or edit
/etc/asterisk/features.conf to move the parking lot.
Either way, there is a fair amount missing from the extensions.conf you
posted. I suggest you troll through http://www.voip-info.org
-Original Message-
From: PHP Mechanic [mailto:[EMAIL PROTECTED]
Sent: September 10, 2004 7:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Call Parking Problem
Hi,
I'm unable to pick up parked calls after they are transfered.
I get the transfer message when I press # and then I'm told
701 The
extension I'm dialing goes to the on hold music. I'm
disconnected, I hang
up, dial 701 and I see this message on the console Everyone is
busy/congested at this time
I just have the default parkedcalls file, and have this in
the extensions.
[AnalogPhone]
exten = _70X,1,Dial(Zap/1/${EXTEN},20,Ttr)
include = parkedcalls
[SipPhone]
exten = _70X,1,Dial(SIP/1/${EXTEN},20,Ttr)
include = parkedcalls
{clip}
___
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: X-SBClass: Spam ( Re: [Asterisk-Users] TDM400P Problems)
I have a problem in that the pci card is not detected at all. Other pci cards are detected fine but not the tdm400b. It's detected on a different - smaller - machine no probs. Any ideas of the magic bios setting to get it going? - Original Message - From: Lyle Giese [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, August 27, 2004 12:41 PM Subject: X-SBClass: Spam ( Re: [Asterisk-Users] TDM400P Problems) Have you checked to see if there is a newer bios for the motherboard? - Original Message - From: Greg Hulands [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, August 26, 2004 9:31 PM Subject: Re: [Asterisk-Users] TDM400P Problems Even when the tdm400 is the only card in the computer, it still has this problem. I'm not sure how I would go about determining what is causing it. I have reset the bios to its factory settings to see if that helped, but alas it did not. Seems like i'm screwed. Greg On 27/08/2004, at 12:12 PM, Lyle Giese wrote: My guess is that you have PCI bus compatibility problems of some sort. Moving the cards around may help. Using a plug in NIC may help. A different Motherboard may help. This looks like hardware and trial and error and experience is all you have to lead you forward, if my guess is right. Lyle - Original Message - From: Greg Hulands [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, August 26, 2004 8:11 PM Subject: Re: [Asterisk-Users] TDM400P Problems When I did the lspci -v when it got to the tdm400p it seemed to go into an infinite loop on this line: Capabilities: [80] #00 []. When I redirected output to a file it filled to 8MB in about 3 seconds with this line. I moved the NIC so that it would be before the tdm400p, but it still did the same thing. I haven't a clue what is going on here. Any help is greatly appreciated. Regards, Greg Here is the output: 00:00.0 Host bridge: nVidia Corporation nForce2 AGP (different version?) (rev c1) Flags: bus master, 66Mhz, fast devsel, latency 0 Memory at d000 (32-bit, prefetchable) Capabilities: [40] AGP version 2.0 Capabilities: [60] #08 [2001] 00:00.1 RAM memory: nVidia Corporation nForce2 Memory Controller 1 (rev c1) Subsystem: Unknown device 17f2:3401 Flags: 66Mhz, fast devsel 00:00.2 RAM memory: nVidia Corporation nForce2 Memory Controller 4 (rev c1) Subsystem: Unknown device 17f2:3401 Flags: 66Mhz, fast devsel 00:00.3 RAM memory: nVidia Corporation nForce2 Memory Controller 3 (rev c1) Subsystem: Unknown device 17f2:3401 Flags: 66Mhz, fast devsel 00:00.4 RAM memory: nVidia Corporation nForce2 Memory Controller 2 (rev c1) Subsystem: Unknown device 17f2:3401 Flags: 66Mhz, fast devsel 00:00.5 RAM memory: nVidia Corporation nForce2 Memory Controller 5 (rev c1) Subsystem: Unknown device 17f2:3401 Flags: 66Mhz, fast devsel 00:01.0 ISA bridge: nVidia Corporation nForce2 ISA Bridge (rev a4) Subsystem: Unknown device 17f2:3401 Flags: bus master, 66Mhz, fast devsel, latency 0 Capabilities: [48] #08 [01e1] 00:01.1 SMBus: nVidia Corporation nForce2 SMBus (MCP) (rev a2) Subsystem: Unknown device 17f2:3401 Flags: 66Mhz, fast devsel, IRQ 12 I/O ports at bc00 Capabilities: [44] Power Management version 2 00:02.0 USB Controller: nVidia Corporation nForce2 USB Controller (rev a4) (prog-if 10 [OHCI]) Subsystem: Unknown device 17f2:3401 Flags: bus master, 66Mhz, fast devsel, latency 0, IRQ 11 Memory at e4002000 (32-bit, non-prefetchable) Capabilities: [44] Power Management version 2 00:02.1 USB Controller: nVidia Corporation nForce2 USB Controller (rev a4) (prog-if 10 [OHCI]) Subsystem: Unknown device 17f2:3401 Flags: bus master, 66Mhz, fast devsel, latency 0, IRQ 9 Memory at e4003000 (32-bit, non-prefetchable) Capabilities: [44] Power Management version 2 00:02.2 USB Controller: nVidia Corporation nForce2 USB Controller (rev a4) (prog-if 20 [EHCI]) Subsystem: Unknown device 17f2:3401 Flags: bus master, 66Mhz, fast devsel, latency 0, IRQ 5 Memory at e4004000 (32-bit, non-prefetchable) Capabilities: [44] #0a [2080] Capabilities: [80] Power Management version 2 00:06.0 Multimedia audio controller: nVidia Corporation nForce2 AC97 Audio Controler (MCP) (rev a1) Subsystem: Unknown device 17f2:3401 Flags: bus master, 66Mhz, fast devsel, latency 0, IRQ 12 I/O ports at c000 I/O ports at b000 [size=128] Memory at e400 (32-bit, non-prefetchable) [size=4K] Capabilities: [44] Power Management version 2 00:08.0 PCI bridge: nVidia Corporation nForce2 External PCI Bridge
[Asterisk-Users] Can't make zaptel on red hat 9
Hello, I've followed the instructions here: http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation, but I get the following when trying to make zaptel: I've added the relevant symlinks: [EMAIL PROTECTED] src]# ls -ld /usr/src/linux* lrwxrwxrwx1 root root 14 Aug 26 22:50 /usr/src/linux - linux-2.4.20-8 lrwxrwxrwx1 root root 14 Aug 27 2004 /usr/src/linux-2.4 - linux-2.4.20-8 drwxr-xr-x 16 root root 4096 Aug 26 22:59 /usr/src/linux-2.4.20-8 How do I get zaptel installed? [EMAIL PROTECTED] zaptel]# make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB - I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-point er -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include /usr/src/linux-2.4/include/linux/modversions.h -DSTANDALONE_ZAPATA -c zaptel.c cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB - I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-point er -I/usr/src/linux/drivers/net/wan -I /usr/src/linux/include -I/usr/src/linux/include/net -DMODVERSIONS -include /usr/src/linux-2.4/include/linux/modversions.h -DSTANDALONE_ZAPATA -c tor2.c In file included from tor2.c:30: /usr/src/linux-2.4/include/linux/kernel.h:60: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:60: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:60: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:61: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:61: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:62: `panic_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:62: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:68: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:68: `simple_strtoul_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:68: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:69: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:69: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:69: `simple_strtol_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:69: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:70: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:70: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:70: `simple_strtoull_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:70: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:72: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:72: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:73: `sprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:73: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:74: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:74: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:74: `vsprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:74: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:75: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:75: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:76: `snprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:76: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:77: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:77: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:77: `vsnprintf_R_ver_str' declared as function returning a function /usr/src/linux-2.4/include/linux/kernel.h:77: warning: function declaration isn't a prototype /usr/src/linux-2.4/include/linux/kernel.h:79: invalid suffix on integer constant /usr/src/linux-2.4/include/linux/kernel.h:79: parse error before numeric constant /usr/src/linux-2.4/include/linux/kernel.h:80: `sscanf_R_ver_str' declared as function returning a function
