[asterisk-users] Asterisk HoneyPot
Hi All, I'm not the first to try to start a VOIP blacklist but currently working on a project for the next 12 hours, hopefully I can get it up soon. What I intend to do is to work with a few reliable Harvester to gather the logs. A simple script to parse it then extract the list of attackers IP, compile them and send them out to the list. If any of you are kind enough to zip and send me a /var/log/asterisk/messages that contain hacker's scan attack, it will be helpful to my research. Do email me at j...@asteriskhoneypot.com . Let me know if you are keen to be a harvester as well.Thanks. Regards, Jackster -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial SIP channel with no registration, timeout before CONGESTION?
What determines how long SIP channel waits, when you dial a peer with no
registration, before returning ${DIALSTATUS} CONGESTION?
When I dial a peer with no registration, SIP channel currently waits
many seconds before returning ${DIALSTATUS} CONGESTION - how can I
shorten this timeout?
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Re: [asterisk-users] Channel failover
On Tue, May 04, 2010 at 08:46:39AM +0100, Steve Howes wrote: On 4 May 2010, at 03:44, Jack Bates wrote: We recently got VoIP, so when we make a call, Asterisk should first try to make the call with VoIP, but in case either our VoIP or our internet service are down, Asterisk should then try to make the call with our old school analog phone line Well, first you try to dial it with the VoIP line.. Then the analogue one... So you just put the two dial commands on separate lines.. Thanks Steve - but wouldn't this mean that when the person we called hangs up, Asterisk will call them again, with the analog phone line? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Channel failover
How do you configure Asterisk to dial, in order, each channel from a group of channels until it either finds an available channel, or runs out of channels? We recently got VoIP, so when we make a call, Asterisk should first try to make the call with VoIP, but in case either our VoIP or our internet service are down, Asterisk should then try to make the call with our old school analog phone line I searched for configuration parameters to put channels in a group, and then dial that group in order until an available channel is found - no luck I also searched the sample configurations for an example, likewise without luck Finally I expected that this topic would be covered in the O'Reilly Asterisk book, http://asteriskdocs.org/ - but again, I found nothing on this topic specifically How do you configure Asterisk to accomplish this? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] directmedia/canreinvite/native bridging question
I've got several SIP clients with dynamic IP addresses Asterisk has one public and one private IP address SIP clients might connect to Asterisk from either the internet or the private network (192.168.1.255) - they're portable By default, directmedia/canreinvite is enabled and Asterisk sets up direct media connections between clients. In this case clients on the internet can make calls between each other, and clients on the private network can make calls between each other, but calls between clients on the internet and clients on the private network suffer from one way audio If I reconfigure Asterisk with directmedia=no then calls between all clients work, but all audio is relayed through Asterisk - even calls between clients in proximity to each other on the internet, but far away from Asterisk : ( I'd like Asterisk to set up direct media connections for calls between clients who're both on the internet, and for calls between clients who're both on the private network, but not set up direct media connections for calls between clients on the internet and clients on the private network I haven't yet figured out how to configure Asterisk to achieve this, and thought I'd ask here if it's possible I looked at directmedia=nonat, but AFAICT enabling this option won't set up direct media connections for calls between clients who're both on the private network? Does Asterisk support what I describe? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] StopPlayTones() after first digit?
I configured our SIP gateway to automatically dial extension s when a phone is picked up. I want Asterisk to play a dial tone, wait for an extension to be dialled, and hangup on timeout This works great, but I also want Asterisk to *stop* playing the dial tone after the first digit is pressed So far my extensions.conf contains, [internal] exten = s,1,Answer exten = s,n,PlayTones(dial) exten = s,n,WaitExten How can I continue playing a dial tone as long as a digit isn't pressed (and TIMEOUT() hasn't expired) - but stop playing the dial tone as soon as the first digit is pressed? I think the Background() application works like this - it stops playing as soon as the first digit is pressed - it seems PlayTones() works differently? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ExitIf() convention?
I want the first line of my dialplan to check and expression, and exit
from the dailplan if it is true - is there a convention for this?
My goal is to exit from the dialplan before calling Answer() if the
callerid is null. By this means I hope to work around this issue:
http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/217794
- I noticed that the callerid is never null for incomming calls - even
blocked numbers are PRIVATE, while on occasions when Asterisk
incorrectly answers during an in progress conversation, the callerid is
null.
Is it correct to use:
exten = s,1,GotoIf(${ISNULL(CALLERID())}?h)
- or is there a more commonly used convention?
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[asterisk-users] anoyingly answers already in use pstn line
I am using Asterisk and an X101P card as a glorified answering machine. We have a residential PSTN line with about six phones connected to it. Like an answering machine, I want Asterisk answer the line *only* when an incoming call is not answered after four rings. This mostly works. My extensions.conf is at the end of this message. The problem is that Asterisk will sometimes answer the line when someone is already talking on one of the six phones connected to it. Sometimes Asterisk will answer the line and start playing the greeting in the middle of a conversation! This is especially a problem when I am talking on the phone to an automated system, because although I hang up the phone I am talking on, neither the automated system nor Asterisk will hang up. I have not yet discovered a pattern to when Asterisk answers the line. It always answers after four rings, but it sometimes answers when someone is already talking on one of the phones connected to the line. In a perfect world, Asterisk would be the only thing connected to the line, and all our phones would be Asterisk extensions. Unfortunately we do not currently have the required VoIP phones or FXS interface... Is there any way to make Asterisk less flaky, and answer the line *only* when an incoming call is not answered after four rings? --- [default] exten = s,1,Wait(20) exten = s,n,Answer exten = s,n,Background(recordings/coop-greeting) exten = s,n(instruct),Background(recordings/leave-message) exten = s,n,Background(recordings/enter-extension) exten = s,n,Background(recordings/dial-by-name) exten = s,n,Background(recordings/visit-website) exten = s,n,WaitExten ; General delivery mailbox exten = #,1,Voicemail(6000) exten = #,n,Goto(s,instruct) ; Dial by name exten = a,1,Directory(default) ; Entering an invalid extension replays the instructions exten = i,1,Playback(invalid) exten = i,n,Goto(s,instruct) ; Timeout goes to voicemail exten = t,1,Goto(#,1) exten = 6003,1,Macro(stdexten,6003,SIP/cstewart) exten = 6004,1,Macro(stdexten,6004,SIP/mhockley) exten = 6005,1,Macro(stdexten,6005,SIP/jbates) [...] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Howto pickup call from queue?
Hi all,
how can I pickup a call from a queue? Which context parameter do I
have to specify? The context that calls the queues application is
ext-queues. This is what I tried so long (777 is the extension of the
queue I want to pickup from):
exten = _**ZXX,1,Noop(Attempt to Pickup ${EXTEN:2} by ${CALLERID(num)})
exten = _**ZXX,n,Pickup(${EXTEN:[EMAIL PROTECTED])
exten = _**ZXX,n,Pickup(${EXTEN:[EMAIL PROTECTED])
exten = _**ZXX,n,Pickup(${EXTEN:2})
exten = _**ZXX,n,Pickup(${EXTEN:[EMAIL PROTECTED])
exten = _**ZXX,n,Pickup(${EXTEN:[EMAIL PROTECTED])
exten = _**ZXX,n,Pickup([EMAIL PROTECTED])
exten = _**ZXX,n,Pickup([EMAIL PROTECTED])
exten = _**ZXX,n,Pickup(777)
exten = _**ZXX,n,Pickup([EMAIL PROTECTED])
exten = _**ZXX,n,Pickup([EMAIL PROTECTED])
This is the CLI output when trying to pickup a call (call from 779 to
queue, agent SIP/758 is ringing):
-- SIP/758-082d3418 is ringing
Extension Changed 779 new state InUse for Notify User 773
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/779-09077e38,
Attempt to Pickup 758 by 779) in new stack
Extension Changed 779 new state InUse for Notify User 776
-- Executing [EMAIL PROTECTED]:2] Pickup(SIP/779-09077e38,
758) in new stack
[Sep 19 11:43:39] NOTICE[12022]: app_directed_pickup.c:159
pickup_exec: No target channel found for 758.
Extension Changed 779 new state InUse for Notify User 758
Extension Changed 779 new state InUse for Notify User 774
-- Executing [EMAIL PROTECTED]:4] Pickup(SIP/779-09077e38,
[EMAIL PROTECTED]) in new stack
[Sep 19 11:43:39] NOTICE[12022]: app_directed_pickup.c:159
pickup_exec: No target channel found for 758.
Extension Changed 779 new state InUse for Notify User 759
-- Executing [EMAIL PROTECTED]:5] Pickup(SIP/779-09077e38,
[EMAIL PROTECTED]) in new stack
[Sep 19 11:43:39] NOTICE[12022]: app_directed_pickup.c:159
pickup_exec: No target channel found for 758.
Extension Changed 779 new state InUse for Notify User 745
-- Executing [EMAIL PROTECTED]:6] Pickup(SIP/779-09077e38,
758) in new stack
[Sep 19 11:43:39] NOTICE[12022]: app_directed_pickup.c:159
pickup_exec: No target channel found for 758.
-- Executing [EMAIL PROTECTED]:7] Pickup(SIP/779-09077e38,
[EMAIL PROTECTED]) in new stack
[Sep 19 11:43:39] NOTICE[12022]: app_directed_pickup.c:159
pickup_exec: No target channel found for 758.
-- Executing [EMAIL PROTECTED]:8] Pickup(SIP/779-09077e38,
[EMAIL PROTECTED]) in new stack
[Sep 19 11:43:39] NOTICE[12022]: app_directed_pickup.c:159
pickup_exec: No target channel found for 758.
-- Executing [EMAIL PROTECTED]:9] Pickup(SIP/779-09077e38,
[EMAIL PROTECTED]) in new stack
[Sep 19 11:43:39] NOTICE[12022]: app_directed_pickup.c:159
pickup_exec: No target channel found for 777.
-- Executing [EMAIL PROTECTED]:10] Pickup(SIP/779-09077e38,
[EMAIL PROTECTED]) in new stack
[Sep 19 11:43:39] NOTICE[12022]: app_directed_pickup.c:159
pickup_exec: No target channel found for 777.
-- Executing [EMAIL PROTECTED]:11] Pickup(SIP/779-09077e38,
777) in new stack
[Sep 19 11:43:39] NOTICE[12022]: app_directed_pickup.c:159
pickup_exec: No target channel found for 777.
-- Executing [EMAIL PROTECTED]:12] Pickup(SIP/779-09077e38,
[EMAIL PROTECTED]) in new stack
[Sep 19 11:43:39] NOTICE[12022]: app_directed_pickup.c:159
pickup_exec: No target channel found for 777.
-- Executing [EMAIL PROTECTED]:13] Pickup(SIP/779-09077e38,
[EMAIL PROTECTED]) in new stack
[Sep 19 11:43:39] NOTICE[12022]: app_directed_pickup.c:159
pickup_exec: No target channel found for 777.
== Auto fallthrough, channel 'SIP/779-09077e38' status is 'UNKNOWN'
Show Channels shows this (call from 201 to queue, SIP/758 is ringing):
asterisk*CLI show channels
Channel Location State Application(Data)
SIP/758-084aa238 (None) Ringing AppQueue((Outgoing Line))
SIP/201-082d83e0 [EMAIL PROTECTED]:11 Up Queue(777|t|||300)
2 active channels
1 active call
Asterisk version is 1.4.10.1 with FreePBX 2.3.0.3.
Thanks for your help.
Regards, Jack
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Re: [asterisk-users] Zaptel channel reservation
Tzafrir Cohen schrieb: On Mon, Jul 30, 2007 at 02:01:49PM +0200, Jack wrote: Hi all, I have a Wildcard TE110P connected to a E1 line an I want to reserve channels in the following way: channels 1-15 and 17-21 for incoming calls channels 22-28 for outgoing calls channels 29-31 for emergency calls My zaptel.conf looks like this: ; incoming group = 1 signalling=pri_cpe context=from-zaptel channel = 1-15 channel = 17-21 ; outgoing group = 2 context = hangup-calls signalling=pri_cpe channel = 22-28 ; emergency group = 3 ; keeping your convention and writing the directive explicitly, ; although it is kept implicitly from previous channel: context = hangup-calls signalling=pri_cpe channel = 29-31 and then in extensions.conf: [hangup-calls] ; not sure that this is precisly the right thing to do: exten = s,1,Hangup This is a solution I was thinking about too, but there is one major problem: When there is a outgoing call, asterisk takes the first available channel, in case there are no active calls this is Zap/22 for outgoing calls in my configuration. If there is a incoming call immediatly after the outgoing call is hangup, asterisk (or the telco?) does not take the first available channel - which would be Zap/1 - it takes Zap/22 instead. So with this solution this incoming call would get lost even when there are no other incoming calls at all. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel channel reservation
Eric ManxPower Wieling schrieb: Jack wrote: Hi all, I have a Wildcard TE110P connected to a E1 line an I want to reserve channels in the following way: channels 1-15 and 17-21 for incoming calls channels 22-28 for outgoing calls channels 29-31 for emergency calls My zaptel.conf looks like this: ; incoming group = 1 signalling=pri_cpe context=from-zaptel channel = 1-15 channel = 17-21 ; outgoing group = 2 signalling=pri_cpe channel = 22-28 ; emergency group = 3 signalling=pri_cpe channel = 29-31 How can I avoid that incoming calls are going to the channels 22-31? You must contact your telco. Actually I preferred to reserve the channels in asterisk, but this seems to be the easiest way. Does anybody know if the mapping from telco channels to zap channels is fixed? Is the first telco channel always mapped to the first zap channel or is this mapping dynamic? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel channel reservation
C F schrieb: Why would you want to do that? let Asterisk (using zap/g in app_dial) take care of which channel are used for outbound but assign all the channels to that g, reject any incoming calls if there are already 7 incoming active calls with a congestion PRI_CAUSE. Do the same for 20 outgoing active calls. That should solve your problem, that is if your problem is that you don't want to allow more than 20 outbound calls, and 7 inbound calls. There should be no need to reserve channels, in fact you could put in much better logic to accomplish what you want efficiently. For example although in general you want 20 outbound calls, but if currently 7 channels are still available (because there are no incoming active calls) you might want to allow up to x additional channels for outbound, until at least 3 incoming channels are active (or whatever the number). The same goes the other way around. Which you wont be able to accomplish by reserving channels. In fact by reserving channels, you might lose lots of incoming calls, while the outbound channels have never reached peak, which basically makes this whole thing a waste of money (on the unused channels). You mention good points and although I (in the moment) have no need to think about wasted channels and wasted money I will keep this in mind. On 7/30/07, Jack [EMAIL PROTECTED] wrote: Hi all, I have a Wildcard TE110P connected to a E1 line an I want to reserve channels in the following way: channels 1-15 and 17-21 for incoming calls channels 22-28 for outgoing calls channels 29-31 for emergency calls My zaptel.conf looks like this: ; incoming group = 1 signalling=pri_cpe context=from-zaptel channel = 1-15 channel = 17-21 ; outgoing group = 2 signalling=pri_cpe channel = 22-28 ; emergency group = 3 signalling=pri_cpe channel = 29-31 How can I avoid that incoming calls are going to the channels 22-31? Regards, Jack ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel channel reservation
Tzafrir Cohen schrieb: On Tue, Jul 31, 2007 at 10:06:30AM +0200, Jack wrote: Tzafrir Cohen schrieb: On Mon, Jul 30, 2007 at 02:01:49PM +0200, Jack wrote: signalling=pri_cpe channel = 29-31 and then in extensions.conf: [hangup-calls] ; not sure that this is precisly the right thing to do: exten = s,1,Hangup This is a solution I was thinking about too, but there is one major problem: When there is a outgoing call, asterisk takes the first available channel , in case there are no active calls this is Zap/22 for outgoing calls in my configuration. If there is a incoming call immediatly after the outgoing call is hangup, asterisk (or the telco?) does not take the first available channel - which would be Zap/1 - it takes Zap/22 instead. So with this solution this incoming call would get lost even when there are no other incoming calls at all. This is because you use Dial(Zap/g3) . Use Dial(Zap/G3) to make asterisk start from the last. You would then need a higher load for calls to be wasted that way. I use Dial(Zap/g2) for outgoing calls. When I change this to Dial(Zap/G2) asterisk uses Zap/28 for the first active outgoing call, but the problem remains the same. When a call comes in a few seconds after the outgoing call is hangup, the call comes in on Zap/28 even if all other channels are not active. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel channel reservation
Hi all, I have a Wildcard TE110P connected to a E1 line an I want to reserve channels in the following way: channels 1-15 and 17-21 for incoming calls channels 22-28 for outgoing calls channels 29-31 for emergency calls My zaptel.conf looks like this: ; incoming group = 1 signalling=pri_cpe context=from-zaptel channel = 1-15 channel = 17-21 ; outgoing group = 2 signalling=pri_cpe channel = 22-28 ; emergency group = 3 signalling=pri_cpe channel = 29-31 How can I avoid that incoming calls are going to the channels 22-31? Regards, Jack ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_features.so / asterisk 1.4.5
Carsten Bock schrieb: José Luis Ledesma schrieb: In my asterisk 1.4.5 chan_features.so has been installed properly... check in your asterisk-source if /channels/chan_features.so is present regards, Jack escribió: Is chan_features.so deprecated for asterisk 1.4.5 or why is this module not installed by asterisk 1.4.5? See the 1.4.5 Changelog: (http://ftp.digium.com/pub/asterisk/ChangeLog-1.4.5) 2007-06-07 19:46 + [r68196] Olle Johansson [EMAIL PROTECTED] * channels/chan_features.c: Disable chan_features by default in menuselect Thanks for your answer and sorry for my late response. So what does this exactly mean to me? Can I keep chan_features.so from 1.4.4? What consequences does it have when chan_features.so is disabled und why has this been done? Is chan_features.so related to features.conf? Regards, Jens ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_features.so / asterisk 1.4.5
Joshua Colp schrieb: Jack wrote: Thanks for your answer and sorry for my late response. So what does this exactly mean to me? Can I keep chan_features.so from 1.4.4? What consequences does it have when chan_features.so is disabled und why has this been done? Is chan_features.so related to features.conf? chan_features.so doesn't provide anything useful, it's not used anywhere because it's not really finished. As for why it was changed to be disabled by default oej probably thought that since we aren't using it why do we have it enabled by default. It's fine to live in a world without chan_features.so :) Thank you very much. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_features.so / asterisk 1.4.5
Hi, after updating from asterisk 1.4.4 to 1.4.5 I get a warning for chan_features.so: Your Asterisk modules directory, located at /usr/lib/asterisk/modules contains modules that were not installed by this version of Asterisk. Please ensure that these modules are compatible with this version before attempting to run Asterisk. Is chan_features.so deprecated for asterisk 1.4.5 or why is this module not installed by asterisk 1.4.5? Regards, Jack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HUDlite Server on Debian Etch/Asterisk 1.4
Hi, has anyone managed to get hudlite server working on a Debian Etch based installation of Asterisk 1.4? So far I managed to eliminate all error messages, but the process is killed directly after starting the hudlite server without showing any error messages. I would be very happy if anyone can give me some hints or point me to a installation guide. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WaitExten not responding on key presses
Hi,
I have the problem that WaitExten is not responding to key presses. Here
are the sections from my extensions.conf:
[globals]
incoming_call=0
menu=0
announce=0
[internal]
exten = 777,1,Goto(hotline,${EXTEN},1)
[hotline]
exten = _X.,1,Set(CALLERID(name)=Hotline)
exten = _X.,n,Set(original_extension=${EXTEN})
exten = _X.,n,GotoIf($[${announce}=1]?4:10)
exten = _X.,n,Answer
exten = _X.,n,NoOp(Ansage: Das Problem XYZ ist bereits bekannt und wird
bearbeitet)
exten = _X.,n,NoOp(Ansage: Druecken Sie die Taste 1 falls Sie wegen
einem anderen Problem anrufen)
exten = _X.,n,NoOp(Ansage: Ansonsten druecken Sie eine andere Taste
oder legen Sie bitte auf)
exten = _X.,n,WaitExten(5)
exten = _X.,n,Goto(18)
exten = _X.,n,Set(menu=1)
exten = _X.,n,NoOp(Ansage: Das Gespraech wird aus Qualitaetsgruenden
aufgezeichnet)
exten = _X.,n,NoOp(Ansage: Falls Sie damit nicht einverstanden sind
druecken Sie bitte die Taste 1)
exten = _X.,n,WaitExten(5)
exten = _X.,n,MixMonitor(test.wav)
exten = _X.,n,SayDigits(123)
exten = _X.,n,Queue(hotline|t|||120)
exten = _X.,n,StopMonitor()
exten = _X.,n,Hangup
exten = _[0-9],1,Goto(menu,${EXTEN},1)
exten = i,1,Goto(invalid,${EXTEN},1)
exten = t,1,Goto(timeout,${EXTEN},1)
[menu]
exten = 1,1,GotoIf($[${menu}=0]?2:4)
exten = 1,n,Set(menu=1)
exten = 1,n,Goto(hotline,${original_extension},11)
exten = 1,n,Goto(hotline,${original_extension},16)
exten = _[02-9*#],1,Hangup
The CLI output is:
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/202-b6d08708, hotline|777|1)
in new stack
-- Goto (hotline,777,1)
-- Executing [EMAIL PROTECTED]:1] Set(SIP/202-b6d08708,
CALLERID(name)=Hotline) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/202-b6d08708,
original_extension=777) in new stack
-- Executing [EMAIL PROTECTED]:3] GotoIf(SIP/202-b6d08708, 0?4:10) in
new stack
-- Goto (hotline,777,10)
-- Executing [EMAIL PROTECTED]:10] Set(SIP/202-b6d08708, menu=1) in
new stack
-- Executing [EMAIL PROTECTED]:11] NoOp(SIP/202-b6d08708, Ansage: Das
Gespraech wird aus Qualitaetsgruenden aufgezeichnet) in new stack
-- Executing [EMAIL PROTECTED]:12] NoOp(SIP/202-b6d08708, Ansage: Falls
Sie damit nicht einverstanden sind druecken Sie bitte die Taste 1) in
new stack
-- Executing [EMAIL PROTECTED]:13] WaitExten(SIP/202-b6d08708, 5) in
new stack
= here is the point where I press a digit but nothing happens:
-- Timeout on SIP/202-b6d08708, continuing...
-- Executing [EMAIL PROTECTED]:14] MixMonitor(SIP/202-b6d08708,
test.wav) in new stack
-- Executing [EMAIL PROTECTED]:15] SayDigits(SIP/202-b6d08708, 123) in
new stack
-- SIP/202-b6d08708 Playing 'digits/1' (language 'de')
== Begin MixMonitor Recording SIP/202-b6d08708
-- SIP/202-b6d08708 Playing 'digits/2' (language 'de')
-- SIP/202-b6d08708 Playing 'digits/3' (language 'de')
-- Executing [EMAIL PROTECTED]:16] Queue(SIP/202-b6d08708,
hotline|t|||120) in new stack
[May 16 11:37:00] WARNING[8400]: translate.c:163 framein: no samples for
alawtolin
-- Started music on hold, class 'default', on SIP/202-b6d08708
-- Stopped music on hold on SIP/202-b6d08708
-- User disconnected from queue hotline while waiting their turn
== Spawn extension (hotline, 777, 16) exited non-zero on
'SIP/202-b6d08708'
== End MixMonitor Recording SIP/202-b6d08708
The real strange thing is that when I change the value of the global
variable announce to 1 WaitExten is working as expected:
[globals]
incoming_call=0
menu=0
announce=1
CLI output:
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/202-081bb9f8, hotline|777|1)
in new stack
-- Goto (hotline,777,1)
-- Executing [EMAIL PROTECTED]:1] Set(SIP/202-081bb9f8,
CALLERID(name)=Hotline) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/202-081bb9f8,
original_extension=777) in new stack
-- Executing [EMAIL PROTECTED]:3] GotoIf(SIP/202-081bb9f8, 1?4:10) in
new stack
-- Goto (hotline,777,4)
-- Executing [EMAIL PROTECTED]:4] Answer(SIP/202-081bb9f8, ) in new stack
-- Executing [EMAIL PROTECTED]:5] NoOp(SIP/202-081bb9f8, Ansage: Das
Problem XYZ ist bereits bekannt und wird bearbeitet) in new stack
-- Executing [EMAIL PROTECTED]:6] NoOp(SIP/202-081bb9f8, Ansage:
Druecken Sie die Taste 1 falls Sie wegen einem anderen Problem anrufen)
in new stack
-- Executing [EMAIL PROTECTED]:7] NoOp(SIP/202-081bb9f8, Ansage:
Ansonsten druecken Sie eine andere Taste oder legen Sie bitte auf) in
new stack
-- Executing [EMAIL PROTECTED]:8] WaitExten(SIP/202-081bb9f8, 5) in
new stack
== CDR updated on SIP/202-081bb9f8
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/202-081bb9f8, menu|1|1) in
new stack
-- Goto (menu,1,1)
-- Executing [EMAIL PROTECTED]:1] GotoIf(SIP/202-081bb9f8, 1?2:4) in new
stack
-- Goto (menu,1,2)
-- Executing [EMAIL PROTECTED]:2] Set(SIP/202-081bb9f8, menu=1) in new
stack
-- Executing [EMAIL PROTECTED]:3] Goto(SIP/202-081bb9f8, hotline|777|11) in
new stack
-- Goto (hotline,777,11)
-- Executing [EMAIL PROTECTED]:11]
[asterisk-users] HUDlite Server on Debian Etch/Asterisk 1.4
Hi, has anyone managed to get hudlite server working on a Debian Etch based installation of Asterisk 1.4? So far I managed to eliminate all error messages, but the process is killed directly after starting the hudlite server without showing any error messages. I would be very happy if anyone can give me some hints or point me to a installation guide. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] background() with m option
Hi...
In my dialplan, I have the following:
exten = s,1,Background(${RECORDING}|m)
exten = s,n,Voicemail(${DID_NO})
exten = 0,1,Voicemail(${DID_NO})
exten = a,1,VoiceMailMain(${DID_NO})
exten = h,1,Hangup
In version 1.2, when I hit 0 during the playback, I will be directed
to voicemail. But in verison 1.4, the call hangs up.
[Jan 24 16:05:37] DTMF[5754]: channel.c:2148 __ast_read: DTMF begin '0'
received on SIP/5060-08c53e68
[Jan 24 16:05:37] DTMF[5754]: channel.c:2128 __ast_read: DTMF end '0'
received on SIP/5060-08c53e68
== Spawn extension (play_recording, s, 1) exited non-zero on
'SIP/5060-08c53e68'
-- Executing [EMAIL PROTECTED]:1] Hangup(SIP/5060-08c53e68, ) in new
stack
== Spawn extension (play_recording, h, 1) exited non-zero on
'SIP/5060-08c53e68'
Does anyone tell me why this is happening?
Thanks,
Jack
___
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Re: [asterisk-users] Hardware TDM Switching (Out Of Office - on vacation)
I will be out the office on vacation. asterisk-users@lists.digium.com 12/15/06 11:25 On Friday 15 December 2006 10:52 am, Eric ManxPower Wieling wrote: DACS is not done in Asterisk. DACS is done in the Zaptel drivers. In fact, you can do DACS without Asterisk even being installed on the system. Also the channels that are DACS'd are not even accessible to Asterisk. You're absolutely right, I keep the two in the same basket in my head. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IVR problem
All, I'm not able to play background files since this morning. I'm seeing this error message in the logs: [Oct 17 10:23:56] WARNING[4572] file.c: File custom/asterisk-prospectus_IVR-main-day does not exist in any format [Oct 17 10:23:56] WARNING[4572] file.c: Unable to open custom/asterisk-prospectus_IVR-main-day (format 0x4 (ulaw)): Permission denied [Oct 17 10:23:56] WARNING[4572] pbx.c: ast_streamfile failed on IAX2/teliax-2 for custom/asterisk-prospectus_IVR-main-day I know the file is there was working last week. I did update some files on the server over the weekend. I built Asterisk from SVN-trunk-r44731. Any help? Thanks, Jack Morgan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR problem
On Tuesday 17 October 2006 11:12, Jack Morgan wrote: All, I'm not able to play background files since this morning. I'm seeing this error message in the logs: [Oct 17 10:23:56] WARNING[4572] file.c: File custom/asterisk-prospectus_IVR-main-day does not exist in any format [Oct 17 10:23:56] WARNING[4572] file.c: Unable to open custom/asterisk-prospectus_IVR-main-day (format 0x4 (ulaw)): Permission denied [Oct 17 10:23:56] WARNING[4572] pbx.c: ast_streamfile failed on IAX2/teliax-2 for custom/asterisk-prospectus_IVR-main-day I know the file is there was working last week. I did update some files on the server over the weekend. I built Asterisk from SVN-trunk-r44731. Any help? Nevermind. Looks like it was a local user permission problem just like the error message indicated. Thanks, Jack Morgan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM22B
I'm having trouble getting my TDM22B to answer a call. I have an analog line plugged into each FXO modules (two analog lines) Neither answer or pickup the call in Astrerisk when dialed from an external phone (eg cell phone). I know the card is working modules zaptel wctdm are loaded. Here is my dmesg output: [17214228.752000] Module 0: Installed -- AUTO FXO (FCC mode) [17214228.952000] Module 1: Installed -- AUTO FXO (FCC mode) [17214229.84] Module 2: Installed -- AUTO FXS/DPO [17214230.728000] Module 3: Installed -- AUTO FXS/DPO [17214230.732000] Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) [17214230.732000] Registered tone zone 0 (United States / North America) [17289961.408000] Registered tone zone 0 (United States / North America) [17291652.28] Registered tone zone 0 (United States / North America) [17291665.348000] Registered tone zone 0 (United States / North America) My /etc/zaptel.conf info: loadzone=us defaultzone=us # Use Kewlstart FXS signalling for the FXO modules 0 and 1 of the TDM422B card fxsks=1-2 # Use Kewlstart FXO signalling for the FXS modules 0 and 1 of the TDM422B card fxoks=3-4 I have two FXO modules in the 1st 2nd slots on the TDM board and two FXS modules in the 3rd 4th lots. My /etc/asterisk/zapata.conf info: [channels] ; defaults ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes restrictcid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=yes ; define channels for FXO modules 0 1 signalling=fxs_ks group=1 channel = 1-2 context=incoming callerid=asreceived ; define channels for FXS modules 0 signalling=fxo_ks channel = 3 context=home callerid=asreceived Any help would be appreciated! Thanks, Jack Morgan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemailmain menu
Hi, Is there way a way to restrict access to certain menus, such as the following: 0 Mailbox options 1 Record your unavailable message 2 Record your busy message 3 Record your name 4 Record your temporary message (new in Asterisk v1.2) Thanks in advance, Jack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk optimizing
Hi, I'm currently running Asterisk 1.2.10 on a dual Xeon 3.4GHz with 2GB RAM and SATA drives, but I'm only able to achieve 120 calls with good audio quality (using G.711u). I'm using realtime for voicemail accounts and ODBC for voicemail storage along with one MySQL when dialing out. The max calls I can achieve is 200 simultaneous but audio is really chopping due to high jitter. Does anyone know how to optimize Asterisk and/or RedHat Enterprise Linux 4 to increase simultaneous calls? Thanks, Jack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk documentation..
http://www.voip-info.org/ look for Asterisk link on the left. Danko Miocevic wrote: Where can I get some asterisk books.. or tutorials..? I´ve been searching in google.. but I find just some tutorials explaining how to fast set up an asterisk server. I want to learn how to use it and how to make my own configurations. So, the thing is that I want to know what is the best book or tutorial that you know? recomendations? Thanks to everyone... Danko Miocevic ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk behind load-balancing switch
Hi, I have 2 SER and 2 Asterisk boxes behind a load-balancing switch. I need Asterisk to initiate the RTP streams to both endpoints. Can that be done? Right now, Asterisk does it part of the time, so I have to create NATs for the Asterisk boxes for the RTP streams to get in. Thanks. Jack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX problem
Hi all, I have 2 servers and I'm trying to configure iax to call from Server2 (fxo) to Server1 (sip extension) Server1: 2 sip's extension (123 and 321) Server2: TDM-400 1 fxo(extension 999, channel 3) and 1 fxs (channel 4) from 123 to 321 it's all right in both ways from 999 to 123 no audio in both ways (the telephone of 123 rings but no sound) Here's my configuration: -- SERVER1 sip.conf - [general] context=sip port=5060 bindaddr=192.168.0.1 disallow=all allow=g729 allow=ulaw language=en [123] type=friend host=dynamic username=123 secret=f599 callerid=123123 context=sip canreinvite=no dtmfmode=rfc2833 mailbox=123 allow=g729 allow=ulaw nat=yes qualify=800 [321] type=friend host=dynamic username=321 secret=f599 callerid=321321 context=sip canreinvite=no dtmfmode=rfc2833 mailbox=321 allow=g729 allow=ulaw nat=yes qualify=800 - SERVER1 extensions.conf [general] static=yes writeprotect=no [default] exten = 123,1,Dial(SIP/123,20,tr) exten = 123,2,Voicemail,u123 exten = 123,102,Voicemail,b123 exten = 321,1,Dial(SIP/321,20,tr) exten = 321,2,Voicemail,u321 exten = 321,102,Voicemail,b321 - SERVER1 iax.conf -- [general] bandwidth=low jitterbuffer=no tos=lowdelay [123] type=user host=192.168.0.2 context=default dtmfmode=rfc2833 --- SERVER2 extension.conf --- exten = 999,1,Dial(IAX2/[EMAIL PROTECTED]/123) no iax.conf at SERVER2 no sip.conf at SERVER2 Debug at SERVER1 SERVER1 iax2 debug IAX2 Debugging Enabled Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 8ms SCall: 1 DCall: 0 [192.168.0.2:4569] VERSION : 2 CALLED NUMBER : 123 CALLING NUMBER : 403 CALLING NAME : Channel 3 LANGUAGE : en USERNAME : 123 FORMAT : 2 CAPABILITY : 65283 ADSICPE : 2 DATE TIME : 189032149 -- Accepting unauthenticated call from 192.168.0.2, requested format = 2, actual format = 2 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACCEPT Timestamp: 7ms SCall: 1 DCall: 1 [192.168.0.2:4569] FORMAT : 2 -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569/1, SIP/123|20|tr) in new stack -- Called 123 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 8ms SCall: 1 DCall: 0 [192.168.0.2:4569] VERSION : 2 CALLED NUMBER : 123 CALLING NUMBER : 403 CALLING NAME : Channel 3 LANGUAGE : en USERNAME : 123 FORMAT : 2 CAPABILITY : 65283 ADSICPE : 2 DATE TIME : 189032149 -- Accepting unauthenticated call from 192.168.0.2, requested format = 2, actual format = 2 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACCEPT Timestamp: 7ms SCall: 1 DCall: 1 [192.168.0.2:4569] FORMAT : 2 -- Executing Dial(IAX2/[EMAIL PROTECTED]:4569/1, SIP/123|20|tr) in new stack -- Called 123 Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: CONTROL Subclass: RINGING Timestamp: 00010ms SCall: 1 DCall: 1 [192.168.0.2:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 7ms SCall: 1 DCall: 1 [192.168.0.2:4569] -- SIP/123-541f is ringing Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: CONTROL Subclass: RINGING Timestamp: 00010ms SCall: 1 DCall: 1 [192.168.0.2:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 7ms SCall: 1 DCall: 1 [192.168.0.2:4569] -- SIP/123-541f is ringing Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 001 Type: CONTROL Subclass: RINGING Timestamp: 00013ms SCall: 1 DCall: 1 [192.168.0.2:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: VOICE Subclass: 2 Timestamp: 00040ms SCall: 1 DCall: 1 [192.168.0.2:4569] Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00040ms SCall: 1 DCall: 1 [192.168.0.2:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00010ms SCall: 1 DCall: 1 [192.168.0.2:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 00013ms SCall: 1 DCall: 1 [192.168.0.2:4569] Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: LAGRQ Timestamp: 10008ms SCall: 1 DCall: 1 [192.168.0.2:4569] Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 003 Type: IAX Subclass: LAGRP Timestamp: 10008ms SCall: 1 DCall: 1 [192.168.0.2:4569] Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 003 Type: IAX Subclass: LAGRQ Timestamp: 10011ms SCall: 1 DCall: 1 [192.168.0.2:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 10008ms SCall: 1 DCall: 1 [192.168.0.2:4569] Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass: LAGRP Timestamp: 10011ms SCall: 1 DCall: 1 [192.168.0.2:4569] Tx-Frame Retry[-01] -- OSeqno: 005 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 10011ms SCall: 1 DCall: 1 [192.168.0.2:4569] -- SIP/123-541f answered
Re: [Asterisk-Users] Problems with zaptel and voice prompts/voicemail
This seems to be due to a driver conflict. If I unload Zaptel, the sound returns. I'm having the same issue with a 2.4 kernel on whitebox 3 using HEAD. Still investigating... let me know if you find anything new. Jack On 6/29/05, Jeremy McDermond [EMAIL PROTECTED] wrote: I've looked all around, and I can't find an answer to this. I apologize if this has been discussed already or is buried somewhere in voip-info.org. I have an asterisk setup on linux 2.6.11.11 kernel, a revision E/F TDM400P, and Polycom IP501 phones. As soon as I load the zaptel module into the kernel, the voice prompts and voicemail system ceases to work. The asterisk logs say that the gsm files are being played, but nothing comes out on the other end. This is for both calls coming in via our VoicePulse Connect lines, or when dialing locally from our SIP phones. As soon as I rmmod the zaptel driver, asterisk acts just fine. Thanks for any assistance the list may be able to provide. -- Jeremy McDermond Xenotropic Systems ___ 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 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] Voicemail and musiconhold sound stopped working
I'm having the same issue. If I unload Zaptel, and restart asterisk... the sound does return. On 7/25/05, Arnd Vehling [EMAIL PROTECTED] wrote: Hi, i am testing stuff for a couple of days now with Asterisk CVS-v1-0-07 and everything worked fine sofar when suddenly the voicemail and musiconhold sound output stopped working. The voicemailmenu still works though. I can see the voiceprompts etc in the debug messages on the asterisk CLI but i cant hear anything. Everything else works fine though. I can call out fine etc. I did some network sniffing using ngrep and verified that the voicemail app is indeed not sending _any_ udp/rtp packets towards my sip fones. I did restore old, working configs back but still no change. I reinstalled asterisk from the cvs and even rebootet my linux box (kernel 2.4.27) still no change. This stuff is now bugging me for 5 hours and i am slowly going nuts. I am using an installation with several different sip-fones, zaptel+zaprtc as well as fcpci+capi on a teles isdn card. Any ideas where to look for? thx, Arnd ___ 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 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] No Sound (2nd post)
I have the same issue, running on WBEL 3.0, I have an older production box that runs fine, but on this new box running the same OS and using the new HEAD I have no audio. I can complete calls just fine, but trying to access any of the system audio yields nothing. Tracking it shows the files are playing and I'm getting no errors so its a bit of a mystery. -Jack On 7/4/05, RockWater ! [EMAIL PROTECTED] wrote: Hello anyone who can help I have two Asterisk boxes with identical hardware (Dev Production). I recently rebuilt the DEV box using Fedora Core 3 and the latest CVS Head. The hardware is an Intel CA810e, onboard everything with a PIII processor. The config is pure VOIP using IAX2 ilBC with Virbiage Firefly soft clients. I also use Ztdummy which seems to be working ok - no error messages. My problem Is that none of the sounds work, there is no sound for any of the following features 1. Voicemail prompts 2. the menu macro in Dial 3. Music on hold 4. conversation Here's everything I have tried so far. 1. update fedora (I have compiled asterisk off the disk release and also after Redhat updates) 2. update Asterisk ( I have recompiled several times over the past month with different HEAD versions) 3. recompile mpg-123 using both 'r' and 'q' versions I am getting a console message from time to time which say Application asterisk uses obsolete OSS audio interface But Dial, Voicemail and Music on hold report no errors. The production system works fine on the older CVS head from Jan 26 2005. With an out of date Fedora install off the CDs. Thanks Craig ___ 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 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] FreeBSD 5.4 (Asterisk 1.0.9) - Playback , MP3Player and Musiconhold not working
I installed Asterisk 1.0.9 in a Freebsd 5.4 ( with no zaptel card); I have 2 zoom x5v and works great ( in extensions 123 and 321 ) but I was trying to test cmd Playback, MusicOnHold, MP3Player but when I call to extension 100 I don't hear the sound ( mp3 or gsm that I put) , I only hear noise If I leave a message in a mailbox the same, all the record is noise - extensionns.conf - [general] static=yes writeprotect=no [sip] exten = 123,1,Dial(SIP/123,20,tr) exten = 123,2,Voicemail,u123 exten = 123,102,Voicemail,b123 exten = 321,1,Dial(SIP/321,20,tr) exten = 321,2,Voicemail,u321 exten = 321,102,Voicemail,b321 exten = 100,1,Answer ;exten = 100,2,MusicOnHold(default) ;exten = 100,2,Playback(tt-weasels,skip) exten = 100,2,MP3Player(/usr/local/share/asterisk/mohmp3/fpm-sunshine.mp3) ;exten = 100,3,Voicemail(100) exten = 100,4,Hangup - I see the mpg123 running /usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s fpm-calm-river.mp3 same happens if I use Playback or MusiconHold. ( I hear noise ) I'm confused, I load the module of the sound card to the kernel.. (this is related to this problem ?? ) + FreeBSD Audio Driver (newpcm) Installed devices: pcm0: VIA VT82C686A at io 0xcc00 irq 10 kld snd_via82c686 (1p/1r/0v channels duplex default) + Any idea or help, will be apreciated. Thanks ___ 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] Leave Message - IVR don't work
I have installed asterisk in a 4.11 RELEASE FreeBSD, and we are using
two Zoom X5v with SIP and works fine, we can call each other and this
is OK
--extensions.conf--
[general]
static=yes
writeprotect=no
[sip]
exten = 123,1,Dial(SIP/123,20)
exten = 123,2,Voicemail(u${EXTEN})
exten = 123,3,Hangup
exten = 123,103,Voicemail(b${EXTEN})
exten = 123,104,Hangup
exten = 321,1,Dial(SIP/321,20)
exten = 321,2,Voicemail(u${EXTEN})
exten = 321,3,Hangup
exten = 321,103,Voicemail(b${EXTEN})
exten = 321,104,Hangup
My problems begins when someone tries to leave a message. Because you
have to guess when to start
to leave the message, I mean that the operator doesn't works ( I set
operator=yes)
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/123/INBOX/msg0002 format: wav49,
0xa9dc200
-- x=1, open writing:
/var/spool/asterisk/voicemail/default/123/INBOX/msg0002 format: gsm,
0xa9dc400
-- x=2, open writing:
/var/spool/asterisk/voicemail/default/123/INBOX/msg0002 format: wav,
0xa9ce400
-- voicemail.conf --
[general]
format=wav49|gsm|wav
servermail=asterisk
operator=yes
attach=yes
maxmessage=180
pbxskip=yes
fromstring=The Asterisk PBX
[default]
123 = 123,digixt,[EMAIL PROTECTED],attach=yes
321 = 321,digixt2,[EMAIL PROTECTED],attach=yes
--
And Music on Hold is not working ( i didn't put that confg up) I only
hear noises if I put it at extensions.conf
musiconhosld.conf
;
; Music on hold class definitions
;
[classes]
default = quietmp3:/usr/local/share/asterisk/mohmp3
loud = mp3:/usr/local/share/asterisk/mohmp3
random = quietmp3:/usr/local/share/asterisk/mohmp3,-z
unbuffered = mp3nb:/usr/local/share/asterisk/mohmp3
quietunbuf = quietmp3nb:/usr/local/share/asterisk/mohmp3
; Note that the custom mode cannot handle escaped parameters
(specifically embedded spaces)
;manual = custom:/usr/local/share/asterisk/mohmp3,/usr/bin/mpg123 -q
-r 8000 -f 8192 -b 2048 --mono -s
---
Really don't know how to debug my sound problems, and of curse how to
continue to try to make IVR work
Thanks for any help!!!
Byes
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[Asterisk-Users] Problems to leave messages in Asterisk
Hi all! I'm using Asterisk as a SIP server with 2 ip phones and it works great, the only think that I can´t make it wotk is: 1. I leave the messages but when I receive them in my mailbox , and open them I hear only noises ---voicemail.conf--- [general] format=wav49 servermail=asterisk attach=yes maxmessage=180 pbxskip=yes fromstring=The Asterisk PBX [default] 123 = 123,peter,[EMAIL PROTECTED],attach=yes 321 = 321,jack,[EMAIL PROTECTED],attach=yes 2. When I try to play an mp3 with MP3player in extensions.conf I only hear noises Any help will be Great!! Thanks ___ 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] Newbie pointers
Hi Kerry, I've been reading your writeups and awaiting each new installment. This project is new to me from your articles, and as my wife will tell you, I have been spending lots of after-hours time working on setting this up since then. :) I suspect you and the link on slashdot have bumped up the trend in interest (not to mention sales of Digium developer kits) quite a bit over the last few weeks. Thanks for you, and to the Asterisk crew for an incredible resource. Kerry Garrison wrote: warning - shameless plug We have been writing some How-To guides and will be doing different product reviews as well. So far, we have had a very good response. Check out our site you will find some things to help get your started. http://www.geekgazette.com -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fred Blaise Sent: Thursday, March 24, 2005 3:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Newbie pointers Hello all I have come to Asterisk with no previous telco experience. As I will be playing with Asterisk really soon, I would like to have some pointers as to some tutorials in telco that could help me get into all this. I am quite a beginner, don't forget :) Thanks a lot! Best, fred ___ 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 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] calling out from a remote * server
You'll want to dial the second server locally using IAX, and have the second
server call out using ZAP
[remote-out]
exten = _8NXXNXX,1,Dial(IAX/x.x.x.x/${EXTEN})
[remote iax] ;in from remote-out
exten = _8NXXNXX,1,Dial(ZAP/g1/${EXTEN:1},70,T)
if you want to call our back and forth... use a different number for
each and if you don't want the area code change the digit pattern to
reflect it... so that after the 8, it fits the pattern or just use _8.
to accept the whole string.
hope this helps,
Jack
On Fri, 15 Oct 2004 21:04:55 +0200 (CEST), Remco Barende
[EMAIL PROTECTED] wrote:
I have set up 2 * servers and connected them via IAX2, the connection
works, so far so good.
To optimize on the phone bill however I would like to have calls
that are local for the remote * server placed through the remote server.
How is this accomplished? I first tried the manual approach (dialing an
8 would make the call go through the remote * server) but it doesn't work,
the call is still placed from the local server.
This is what I put in my extensions.conf:
[remote-out]
switch = IAX2/user:[EMAIL PROTECTED]/pstn-local
exten = _8.,1,Dial(ZAP/g1/${EXTEN:1},70,T)
exten = _1NXXNXX,1,Dial(ZAP/g1/${EXTEN})
exten = _NXX,1,Dial(ZAP/g1/${EXTEN})
exten = _8.,2,Macro(fastbusy)
Ideally I would also like * to strip the area code if the remote server is
used (it's a local call then) but this is detail. Ultimately I would like
to do the same with international calls.
I couldn't find the solution in the wikis.
Thanks all!
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[Asterisk-Users] Automated calling/Bridging and takedown in Asterisk?
I am working on a web phone interface to give normal phonesets more 'virtual buttons'..etc, like the expensive executive phones via control via the web. This lead me to the following issue: I am wondering if it is possible (it doesn't seem to as far as I can tell) to make a script (AGI or otherwise) that will have asterisk automatically do the following without the user needing to originate any calls on their telephone: -Call an extension, hold on to the call (call A) -Call another extension, hold on to the call (call B) -Bridge the two calls (A and B) together (so the two extensions can talk to each other) -Later the script drops call B, but keeps call A up -Then asterisk calls another extension (call C) -Then asterisk bridges A with C so then they can talk to each other ..then later the same thing again (call D, then bridge with A)..etc...etc, Allthis would be AGI or script driven without any user having to press anything on his phone. Is this possible (the main issue I see in asterisk is that I cannot find a command in the asterisk API to bridge/unbridge calls like this without something being originated by a call into asterisk from a user). I looked at meetme, but it doesn't seem appropriate for what I want to do above. Any ideas? Thank you! Jack ___ Do you Yahoo!? Express yourself with Y! Messenger! Free. Download now. http://messenger.yahoo.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] ztdummy running, but moh meetme don't work
Any thoughts on the following? I am running asterisk from CVS (downloaded yesterday's version, just to be sure) on a test system with no digium cards in it, so I have installed ztdummy (see logs and screenshots below) as a timing source. When I call the music on hold extension from a Sipura Sip connected analog phone, I hear nothing and start getting Warning[98310]: chan_sip.c:674 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Response) As well, I set up a meetme conference, and dial it, the first user (also a Sipura sip phone) gets 'there are no other users on the conference.., which is OK, then a second user comes in, but they are not conferenced anymore. I can hang up both phones, and dial back to the conference, but I won't even hear the 'there are no other users message anymore'. usb-uhci and ztdummy are loaded fine (see lsmod), and this system is running Redhat9 standard install with linux sources. Any thoughts what might be wrong? I have already spent the whole night googling and looking around, so I think I covered all the basics already. I tried to use zaptelrtc as an alternative to ztdummy, but it doesn't compile on redhat9 (log below as well), so that is not an alternative either. Is ztdummy fairly reliable, or does it not work on some motherboard usb chipsets? (this is a compaq deskpro pentium 400mhz) Is there something I need to do with my kernel (recompile?) so that ztdummy works, or anything else. (I suspect the cause is ztdummy, since both MOH and Meetme are broken..) Thank you --- Logs/Listings #service zaptel start Loading zaptel framework: [ OK ] Loading zaptel hardware modules: wcusb Running ztcfg: [ OK ] #modprobe ztdummy --lsmod listing #lsmod Module Size Used byNot tainted soundcore 6116 0 (autoclean) ztdummy 2532 0 (unused) parport_pc 17508 1 (autoclean) lp 8580 0 (autoclean) parport33952 1 (autoclean) [parport_pc lp] iptable_filter 2316 0 (autoclean) (unused) ip_tables 14488 1 [iptable_filter] autofs 12148 0 (autoclean) (unused) e100 56644 1 wcusb 20064 0 (unused) zaptel179840 4 [ztdummy wcusb] keybdev 2720 0 (unused) mousedev5204 0 hid20772 0 (unused) input 5632 0 [keybdev mousedev hid] usb-uhci 24652 0 [ztdummy] usbcore73088 1 [wcusb hid usb-uhci] ext3 64704 2 jbd47828 2 [ext3] --extensions.conf (relavent part) ;dial 500 to join the conference (doesn't work though) exten=500,1,Answer exten=500,2,MeetMe(1234) ... ;dial 6000 to hear music on hold (doesn't work though) exten = 6000,1,Answer exten = 6000,2,MusicOnHold,default --Meetme.conf [rooms] ; ; Usage is conf = confno[,pin] ; conf = 1234 --musiconhold.conf [classes] default = quietmp3:/var/lib/asterisk/mohmp3 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] Asterisk support for Japanese telephone system?
Does any combination of Asterisk hardware and software exist that supports connection to the Japanese telephone system? (T1, E1 or J1 would be preferable, but analog would be OK as well as a last resort (20 lines though..)). Anyone have any thoughts or experience with this, (or if Asterisk is just not ready for Japan just yet?) Please let me know, Thank you! __ Do you Yahoo!? Yahoo! Mail - More reliable, more storage, less spam http://mail.yahoo.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] No field 'Via' present to copy
Hi I wonder if anyone can throw some light on the * console message. This only occurs when I register a phone on the end of a BT ADSL line, with a Draytec router/modem. The phone registers okay but cannot dial out. Console message: Notice[1125329600]: File chan_sip.c, Line 1759 (copy_via_headers): No field Via present to copy Thanks Steven * Steven Jack University of Glasgow Computing Service Glasgow G12 8QQ [EMAIL PROTECTED] Tel +44(0)141 330 3828 Fax +44(0)141 330 3820
