[asterisk-users] Asterisk outbound register messages causes firewall issues
Hello! I have an Asterisk box that has multiple (over 50) individual accounts registering against Ericsson IMS pbx. Problem is, that when there’s a problem in the network, asterisk will retry to register all the accounts at once, causing the IMS to think it is an attack and block the ip. Asterisk continues to bombard the IMS and the block contionues to be applied. To get out of the loop, blocking service on the IMS must be manuallt disabled for some time to allow asterisk to register, then everything works until next time. My question is, is there any way to make asterisk NOT send all registers at once? Or change source port per account? What is the possible solution, if there even is one? Thanks in advance Rennes Neps kõnesidelahenduste ekspert / Voice Services Solutions Expert Elion Ettevõtted AS tel: +372 6402183 mob: +372 56490388 rennes.n...@elion.eemailto:rennes.n...@elion.ee From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chandrakant Solanki Sent: Tuesday, September 04, 2012 9:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Interrupt error Hello, Asterisk : asterisk-1.6.0.5 Dahdi: dahdi-linux-complete-2.5.1 Kernel Version: 2.6.18-128.el5xen AS_1 kernel: Uhhuh. NMI received for unknown reason 00 on CPU 0. Message from syslogd@ at Tue Sep 4 11:46:57 2012 ... AS_1 kernel: Do you have a strange power saving mode enabled? Message from syslogd@ at Tue Sep 4 11:46:57 2012 ... AS_1 kernel: Dazed and confused, but trying to continue Message from syslogd@ at Tue Sep 4 11:49:39 2012 ... AS_1 kernel: Uhhuh. NMI received for unknown reason 00 on CPU 0. Message from syslogd@ at Tue Sep 4 11:49:39 2012 ... AS_1 kernel: Do you have a strange power saving mode enabled? Message from syslogd@ at Tue Sep 4 11:49:39 2012 ... AS_1 kernel: Dazed and confused, but trying to continue Message from syslogd@ at Tue Sep 4 11:52:17 2012 ... AS_1 kernel: Uhhuh. NMI received for unknown reason 00 on CPU 0. Message from syslogd@ at Tue Sep 4 11:52:17 2012 ... AS_1 kernel: Do you have a strange power saving mode enabled? Message from syslogd@ at Tue Sep 4 11:52:17 2012 ... AS_1 kernel: Dazed and confused, but trying to continue Message from syslogd@ at Tue Sep 4 11:52:27 2012 ... AS_1 kernel: Uhhuh. NMI received for unknown reason 00 on CPU 0. Message from syslogd@ at Tue Sep 4 11:52:27 2012 ... AS_1 kernel: Do you have a strange power saving mode enabled? Message from syslogd@ at Tue Sep 4 11:52:27 2012 ... AS_1 kernel: Dazed and confused, but trying to continue -- Regards, Chandrakant Solanki -- _ -- 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
Re: [asterisk-users] fallback to default extension
Hey,
I would also recommend to use SIPPEER and with that verify the status of said
peer. Based on that status, make the dialling decision.
If you want more help, contact me directly.
Rennes Neps
Elion Ettevõtted AS
tel: +372 6402183
mob: +372 56490388
rennes.n...@elion.ee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Frost
Sent: Wednesday, March 21, 2012 2:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] fallback to default extension
On Mar 21, 2012, at 08:36 , Andrew Latham wrote:
On Wed, Mar 21, 2012 at 8:27 AM, Paolo Supino paolo.sup...@gmail.com wrote:
Hi
I was asked by our development departement to setup asterisk in a
manner that if someone calls an extension in the department that was
was only configured, but a handset was never attached to it to fall
back to a default extension. For example: Someone calls extension
2408, but there's no phone attached to 2408 it should fall back and
ring at 2400..
How do I setup asterisk to find out if there's a phone attached to an
internal number if not ring another extension?
Just add a dial(SIP/2400) at a later priority or any of the other many
ways. Assuming 2400 is you operator then set the var and drop to the
operator. Verify your options to you dial syntax and any std-exten
setups.
You might want to additionally inspect ${DIALSTATUS} to know more about why the
first Dial() (to 2408, in your example) failed, and then use the ExecIf or
GotoIf applications to take different actions.
You might also try the function SIPPEER, again coupled with ExecIF or GotoIf.
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Re: [asterisk-users] How can I get SIP/SDP values to a variable?
Thanks for the quick reply, I was afraid of that. Oh well. :) Rennes Neps Elion Ettevõtted AS rennes.n...@elion.ee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Tuesday, March 13, 2012 1:31 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How can I get SIP/SDP values to a variable? On 03/13/2012 05:58 AM, Rennes Neps wrote: I wonder if there's a way to read SDP values into a variable in Asterisk? I have successfully used SIP_HEADER function to get all I want out of SIP part of the message, no problem. But I would like to be able to read SDP part as well, anyone? Reason I want to to this is: testing the SIP_ALG condition in incoming invite message. Majority of cases can be detected just by comparing src ip address and via, but some devices only rewrite SDP c value for example and so on ... I haven't found any information anywhere how to achieve reading SDP with asterisk. I can use 1.8 and 10 versions. There is no mechanism in Asterisk to do this. The most practical approach would probably be to put a stateless SIP proxy in front of Asterisk and write whatever logic you like there, causing it to add one or more headers to the SIP messages that can be accessed from the dialplan in Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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 -- _ -- 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
Re: [asterisk-users] ChanSpy
No, Extenspy was introduced in 1.4 as far as I know.
Chanspy is simple :) Helpful as I am, I'm gonna paste here the output of show
application chanspy
callcenter*CLI show application ChanSpy
callcenter*CLI
-= Info about application 'ChanSpy' =-
[Synopsis]
Listen to a channel, and optionally whisper into it
[Description]
ChanSpy([chanprefix][|options]): This application is used to listen to the
audio from an Asterisk channel. This includes the audio coming in and
out of the channel being spied on. If the 'chanprefix' parameter is specified,
only channels beginning with this string will be spied upon.
While spying, the following actions may be performed:
- Dialing # cycles the volume level.
- Dialing * will stop spying and look for another channel to spy on.
- Dialing a series of digits followed by # builds a channel name to append
to 'chanprefix'. For example, executing ChanSpy(Agent) and then dialing
the digits '1234#' while spying will begin spying on the channel
'Agent/1234'.
Options:
b - Only spy on channels involved in a bridged call.
g(grp)- Match only channels where their ${SPYGROUP} variable is set
to
contain 'grp' in an optional : delimited list.
q - Don't play a beep when beginning to spy on a channel, or
speak the
selected channel name.
r[(basename)] - Record the session to the monitor spool directory. An
optional base for the filename may be specified. The
default is 'chanspy'.
v([value])- Adjust the initial volume in the range from -4 to 4. A
negative value refers to a quieter setting.
w - Enable 'whisper' mode, so the spying channel can talk to
the spied-on channel.
W - Enable 'private whisper' mode, so the spying channel can
talk to the spied-on channel but cannot listen to that
channel.
Hope you get it to work. Also http://voip-info.org is a great source of
information about asterisk. Good luck
Regards
Rennes Neps
-Original Message-
From: asterisk-users-boun...@lists.digium.com on behalf of Torintino T
Sent: Wed 10/14/2009 22:48
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] ChanSpy
Thanks for your reply.
Is ExtenSpy available in Asterisk 1.2?
If yes, please how can i use it?
and how can i cycle through the available channels by ChanSpy?
Thanks.
Torintino
Date: Wed, 14 Oct 2009 18:15:29 +0300
From: rennes.n...@norby.ee
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] ChanSpy
You must use extenspy if you want to spy on specific extension. Otherwise you
can only cycle through available channels.
Regards
Rennes
-Original Message-
From: asterisk-users-boun...@lists.digium.com on behalf of Torintino T
Sent: Wed 10/14/2009 17:46
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ChanSpy
I am unsing Asterisk 1.2.28
I want please to use ChanSpy urgently
my /etc/asterisk/extensions_additional.conf is as follow:
[chanspy]
include = chanspy-custom
exten = 102**,1,Chanspy(102)
exten = 102**,n,Hangup
exten = 103**,1,Chanspy(103)
exten = 103**,n,Hangup
exten = 400**,1,Chanspy(400)
exten = 400**,n,Hangup
exten = 501**,1,Chanspy(501)
exten = 501**,n,Hangup
exten = 601**,1,Chanspy(601)
exten = 601**,n,Hangup
exten = 606**,1,Chanspy(606)
exten = 606**,n,Hangup
; end of [chanspy]
I created a Context to put my extension into it to be able to use ChanSpy.
While there is a call with an extension 102 and my extension is 606
i call 102** to spy but i couldn't hear anything, all i hear is beep
-- Executing ChanSpy(SIP/606-09430fd0, 102) in new stack
-- Playing 'beep' (language 'en')
-- Playing 'beep' (language 'en')
Thanks
Torintino
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Re: [asterisk-users] ChanSpy
You must use extenspy if you want to spy on specific extension. Otherwise you can only cycle through available channels. Regards Rennes -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Torintino T Sent: Wed 10/14/2009 17:46 To: asterisk-users@lists.digium.com Subject: [asterisk-users] ChanSpy I am unsing Asterisk 1.2.28 I want please to use ChanSpy urgently my /etc/asterisk/extensions_additional.conf is as follow: [chanspy] include = chanspy-custom exten = 102**,1,Chanspy(102) exten = 102**,n,Hangup exten = 103**,1,Chanspy(103) exten = 103**,n,Hangup exten = 400**,1,Chanspy(400) exten = 400**,n,Hangup exten = 501**,1,Chanspy(501) exten = 501**,n,Hangup exten = 601**,1,Chanspy(601) exten = 601**,n,Hangup exten = 606**,1,Chanspy(606) exten = 606**,n,Hangup ; end of [chanspy] I created a Context to put my extension into it to be able to use ChanSpy. While there is a call with an extension 102 and my extension is 606 i call 102** to spy but i couldn't hear anything, all i hear is beep -- Executing ChanSpy(SIP/606-09430fd0, 102) in new stack -- Playing 'beep' (language 'en') -- Playing 'beep' (language 'en') Thanks Torintino _ Windows Live: Make it easier for your friends to see what you're up to on Facebook. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.5.421 / Virus Database: 270.13.112/2391 - Release Date: 10/13/09 19:11:00 winmail.dat___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] libss7 problem with dialing a non numeric string
Hei! I'm trying to send special characters out to ss7 link, but libss7 seems to convert them to zeroes. The challenge is that our service provider demands some of the regional numbers to be sent in format D0+number. When I use D in front of the number in dialplan, libss7 replaces it with 00, So I have a dial string: exten = _[A-Z].,1,Dial(DAHDI/g1/DD0501,,g) But in SS7 trace I see: --VARIABLE LENGTH PARMS[1]-- Called Party Number: Nature of address: 3 NI: 0 Numbering plan: 1 Address signals: 00501# [ 07 03 10 00 00 00 05 f1 ] Do you have any idea how to fix that? My chan_dahci.conf is as follows: [channels] switchtype=euroisdn ;;; linkset 1 context=incoming_ss7 echocancel=yes echotraining=yes echocancelwhenbridged=yes group=1 linkset=1 signalling=ss7 ss7type=itu pointcode=50 adjpointcode=14 defaultdpc=14 ;networkindicator=international networkindicator=national_spare ss7_called_nai=dynamic ss7_calling_nai=dynamic ;ISDN call type ss7_internationalprefix = 00 ;ss7_nationalprefix = ;ss7_subscriberprefix = ;ss7_unknownprefix = DD cicbeginswith = 2 channel = 2-31 sigchan=1 Thanks in advance BR Rennes Neps ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there a way to get info who disconnected the call into CDR?
Hei! Here's my problem. I have an Asterisk with SS7 and SIP trunks. Asterisk version is 1.6. I'm setting up a custom CDR fields and I was wondering is there a way to know who initiated a hangup? Asterisk must be aware of that info somehow, cause in queue_log, that info is present (completecaller, completeagent) Is there a way to get that info on the regular SS7 to SIP (and vica versa) calls? Best regards Rennes Neps ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to get info who disconnected thecall into CDR?
Found it, I use the g flag in Dial command, that helps :) Rennes From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rennes Neps Sent: 1. oktoober 2009. a. 16:05 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Is there a way to get info who disconnected thecall into CDR? Hei! Here’s my problem. I have an Asterisk with SS7 and SIP trunks. Asterisk version is 1.6. I’m setting up a custom CDR fields and I was wondering is there a way to know who initiated a hangup? Asterisk must be aware of that info somehow, cause in queue_log, that info is present (completecaller, completeagent) Is there a way to get that info on the regular SS7 to SIP (and vica versa) calls? Best regards Rennes Neps No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.5.409 / Virus Database: 270.13.112/2391 - Release Date: 09/30/09 18:56:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to measure call lenght and act upon it?
Hei! I need to accomplish something and I don't know how. Asterisk version is 1.2.13. I need to make a routing decision based on how long the call has been in ringing state. Lets say I have a few extensions and I want to ring each of them for 5 seconds (I can't use queue for technical reasons) Of course when some exten is busy it will try the next and so on. But the difficult part is, I want to play a file when call has been ringing for 10 seconds. No matter where in the routing the call may be ringing at that moment. There for I can't just play the file inside the routing. Is there some counter or variable which I could check before each next hop? Or do I have to do the math based on timestamp? That could be messy ... for me at least. Best regards Rennes Neps Norby Telecom No virus found in this outgoing message. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.6.10/1638 - Release Date: 27.08.2008 19:06 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Start call from asterisk
Hint: look into asterisk call files. Create call files with PHP or whatever and drop them into asterisk's outbound calls directory. There's plenty of samples on voip-info.org Regards Rennes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Suity Zsolt Sent: Friday, October 26, 2007 3:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Start call from asterisk Hi everybody! Can you give me a hint, how can I start a call from asterisk with some (php, bash, etc) script? I need to start two calls and bride it together. Thank you. -- Suich ___ --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] Sipura SPA-841 autodial?
I looked over and over again and I still can't find it. Can you please point it out to me? Maybe I'm just looking for the wrong term or something. Thanks in advance Rennes Neps Eric Wieling wrote: Rennes Neps wrote: Hei! Does anyone know how to configure this phone to autodial the number after interdigit timeout has passed? It's documented on the SIPura web site and the various documentation for other SIPura products. However, with a proper dialplan in the phone you seldom need to deal with a timeout as the phone will dial the number as soon at it gets a unique match. ___ 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 Spam detection software, running on the system zeus.avanzada7.com, has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Rennes Neps wrote: Hei! Does anyone know how to configure this phone to autodial the number after interdigit timeout has passed? It's documented on the SIPura web site and the various documentation for other SIPura products. However, with a proper dialplan in the phone you seldom need to deal with a timeout as the phone will dial the number as soon at it gets a unique match. 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 [...] Content analysis details: (0.1 points, 5.0 required) pts rule name description -- -- 0.1 FORGED_RCVD_HELO Received: contains a forged HELO ___ 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] Sipura SPA-841 autodial?
Hei! Does anyone know how to configure this phone to autodial the number after interdigit timeout has passed? Rennes ___ 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] DTMF Payload type
Hei!! 101 type in Snom is correct. Don't change that. Rennes Michael Di Martino wrote: To All I am using a SNOM 190 w/Asterisk server. Here is my sip.conf [7501] type=friend context=external username=7501 callerid=Telx 7501 7501 [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 My question is this. With above settings in my sip.conf specially dtmfmode=rfc2833 What should my DTMF Payload Type: be set to on my SNOM 190 phone. Currently it is set to 101. Should it be set to rfc2833? Regards, Michael DiMartino ___ 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] Door buzzer.
I have the same problem setup with one of our customers, but I have a different problem. I have Grandstream ATA 486 connected to clients doorphone system and clients *. I have two problems: first, when someone is calling from the doorphone, ATA doesn't recognize the called number correctly. 2 or tries out of 10, ATA get's the number wrong. I have tried all kinds of DTMF settings, relax dtmf and so on, nothing helps. It seems to me, the doorphone's generated DTMF tones are too short. Ok, that I can resolve with some simple hack, but bigger problem is, when secretary presses 8 on the phone, to open the door, doorphone doesn't recognize the tone. Customer has SNOM 190's and BT-100 on their network. Now however long I press the button on the phone, * still sends a very short tone on the line. And that doesn't seem to enough for the doorphone to recognize. Is there any way to make * generate longer DTMF tones? Regards Rennes Neps Cian O'Sullivan wrote: Hello, I have a customer who has their front door integrated to their current phone system. If someone presses the buzzer, the secretaries phone will ring, and she can talk to the person at the door. By pressing ** she can release the door. Anyone have any sort of integration like this. Are there IP devices anyone is using? They have a pizza box server as their asterisk server with a T1 card. No more slots, so if I want to use the existing infrastructure I will need to build a second server with an FXO port. Kinda stupid having a second server just to open the door. Any suggestions? Cian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Door buzzer.
I have the same problem setup with one of our customers, but I have a different problem. I have Grandstream ATA 486 connected to clients doorphone system and clients *. I have two problems: first, when someone is calling from the doorphone, ATA doesn't recognize the called number correctly. 2 or tries out of 10, ATA get's the number wrong. I have tried all kinds of DTMF settings, relax dtmf and so on, nothing helps. It seems to me, the doorphone's generated DTMF tones are too short. Ok, that I can resolve with some simple hack, but bigger problem is, when secretary presses 8 on the phone, to open the door, doorphone doesn't recognize the tone. Customer has SNOM 190's and BT-100 on their network. Now however long I press the button on the phone, * still sends a very short tone on the line. And that doesn't seem to enough for the doorphone to recognize. Is there any way to make * generate longer DTMF tones? Regards Rennes Neps Cian O'Sullivan wrote: Hello, I have a customer who has their front door integrated to their current phone system. If someone presses the buzzer, the secretaries phone will ring, and she can talk to the person at the door. By pressing ** she can release the door. Anyone have any sort of integration like this. Are there IP devices anyone is using? They have a pizza box server as their asterisk server with a T1 card. No more slots, so if I want to use the existing infrastructure I will need to build a second server with an FXO port. Kinda stupid having a second server just to open the door. Any suggestions? Cian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Time announcement
Hei! Should be something like this: exten = exten_number,1,Answer exten = exten_number,2,DateTime() exten = exten_number,3,Dial(SIP/exten_num,30,) Your application may vary... Rennes Ronald Wiplinger wrote: I would like to let my callers know what time it is before I switch them to an extension number. (They should know that it is 3 am in the morning, when they are calling me) Is there such an application available? bye Ronald ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can I trigger an application in * with DTMF tones, during a call?
I had an idea, to use user input DTMF tones as a trigger to start recording a conversation ... Haven't found any examples on it though ... Any thoughts? Regards Rennes Neps ___ 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] Problems with MACRO_EXTEN variable
Hei!
I have a little problem with the subject. I use Asterisk
CVS-HEAD-09/06/04-12:42:56 as a production *, but I do tests with a
newer version
Asterisk CVS-HEAD-11/18/04-10:01:32. Ok the problem is:
in extension.conf I use macro for redirection, found on wiki pages:
[macro-stdexten]
;
; Standard extension macro (with call forwarding):
; ${ARG1} - Extension(we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten=s,1,DBget(temp=CFIM/${ARG1}) ; Get CFIM key, if not existing, goto 102
exten=s,2,SetCallerID(${MACRO_EXTEN})
exten=s,3,Dial(Local/[EMAIL PROTECTED]/n) ; Unconditional forward
exten=s,4,Dial(${ARG2},40,Tt) ; 20sec timeout
exten=s,5,DBget(temp=CFBS/${ARG1}) ; Get CFBS key, if not existing,
goto 105
exten=s,6,SetCallerID(${CALLERIDNUM})
exten=s,7,Dial(Local/[EMAIL PROTECTED]/n) ; Forward when busy or unavailable
exten=s,8,Dial(${ARG2},40,Tt) ; 20sec timeout
;exten=s,9,DBget(temp=CFNA/${ARG1}) ; Get CFNA key, if not existing,
goto 109
;exten=s,10,SetCallerID(${CALLERIDNUM})
;exten=s,11,Dial(Local/[EMAIL PROTECTED]/n) ; Forward when busy or
unavailable
;exten=s,12,Dial(${ARG2},20) ; 20sec timeout
; No CFIM key
exten=s,102,Goto(s,5)
; No CFBS key
;exten=s,105,Goto(s,9)
; No CFNA key - voicemail ?
;exten=s,109,(Busy)
[apps]
; Unconditional Call Forward
exten = _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4})
exten = _*21*X.,2,Hangup
exten = #21#,1,DBdel(CFIM/${CALLERIDNUM})
exten = #21#,2,Hangup
; Call Forward on Busy or Unavailable
exten = _*67*X.,1,DBput(CFBS/${CALLERIDNUM}=${EXTEN:4})
exten = _*67*X.,2,Hangup
exten = #67#,1,DBdel(CFBS/${CALLERIDNUM})
exten = #67#,2,Hangup
On the older version everything works fine, MACRO_EXTEN changes the
callerid of a redirected call as it is supposed to. But with newer
version, it messes up the name part of the CALLER_ID. On Snom phones,
that show number and name, total crap is displayed instead of name. Same
crap gets put into MYSQL database into CLID column. * cli debug shows
nothing weird. CVS-HEAD-11/18/04-10:01:32 Changelog says - -- Major
Caller*ID Restructuring, but doesn't explain the details. Can anyone help?
Regards
Rennes Neps
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