[Asterisk-Users] Bad soundquality on inbound calls.
Hi all, This is my first question to this list, so please be gentle... Last week I installed a X100P FXO-card. And with not much tweaking I had it running fine, there is only one problem right now and it is the soundquality. When making a call sound is always perfect, both for the calling party and the called party. The problem is recieving phonecalls. The soundquality differs a lot then. The near user hears the far-user very low. And the far user says the sound is really bad, with a lot of distortion. Anyone recognize this type of problem and what to do about it? Or maybe someone can just point me in the right direction for bugfixing this. (Ive read alot about zapata.conf and zaptel.conf. But none of the things i've found have helped) My setup is a 2,6GHz P4 with 256MB RAM running Fedora Core 3. All phones are software phones (X-lite) using SIP to register on the server, so it would look somethnk like this. IP Phone - Asterisk - PSTN Kind Regards, Jakob ### Detta mail har blivit skannat av F-Secure Anti-virus for Microsoft Exchange. For mer information, ga till http://www.F-Secure.se This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.F-Secure.com/ ___ 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
AW: [Asterisk-Users] Transfer a call ? Am I lookingfortheflashcommand ?
Hello Jim,
I tryed that with capi.. but no luke. It will hang up the line anyway :-(
exten = s,1,Playback(transfer)
exten = s,2,Flash(capi/72044**:041720,18)
exten = s,3,SendDTMF(${ARG1})
exten = s,4,Hangup()
Any idears why ?
BTW: Whats actually that SendDTMF ? thing ?
Thx for the help..
Grüsse / Best Regards
Mateo Meier
-
Don't marry for money; you can borrow it cheaper ;-)
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Jim Van
Meggelen
Gesendet: Samstag, 26. Februar 2005 07:54
An: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion'
Betreff: RE: [Asterisk-Users] Transfer a call ? Am I
lookingfortheflashcommand ?
[EMAIL PROTECTED] wrote:
Hello Jim,
thx for the answer..
Im happy I found someone that is using flash :)
It's not perfect, but it can be useful.
Am I right, if I transfer a call with flash, the line will be free
afterwards ?
Yep
Would you mind to past me how you did the flash part @the
extention file ? Also, If I use flash, do I have to setup
anything else or just @the extention file ?
Jere's the relevant section of my dial plan:
[macro-cell_user]
exten = s,1,Playback(transfer)
exten = s,2,Flash(zap/1)
exten = s,3,SendDTMF(${ARG1})
exten = s,4,Hangup()
Good luck!
Jim.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Jim
Van Meggelen Gesendet: Freitag, 25. Februar 2005 05:57
An: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Betreff: RE: [Asterisk-Users] Transfer a call ? Am I looking for
theflashcommand ?
[EMAIL PROTECTED] wrote:
On Fri, 2005-02-25 at 00:50 +0100, Mateo Meier wrote:
Hey Guys
Im trying to forward a call with asterisk to a regular phone.
Something like I get a call on my regular phone, and he's trying
to reach some buddy of mine.. then I tell him wait a sec and push
Flash and get a other dialtone.. then I dial that other number
then hangup the phone, so the one that called will be connected to
where I dialed it to...
Some buddy of mine told me im looking for a function called flash
Only thing Im able to find is:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Flash
Im unsure how to use it now..
Let's say if I forward a call with asterisk as following: exten =
2,1,Dial(capi/720:07812345*,18)
How would I use the flash command to transfer that call above to 078
12345* ? I have no problem transferring a call, but when Im doing
this with the dial command (see above).. then my line will be busy
Been covered before, You can't do that on an analog line. Problem
comes from where you are and what flash would be working on at that
point. If you flash asterisk and get dialtone again, you are getting
the dialtone
from asterisk. At this point the only channel being worked is the one
you are on and flashing it won't help.
What you would need to do is get the other leg of the call to make
the flash.
It might be really handy to be able to specify the trunk to
flash() as an argument. I use flash in my dialplan to
transfer incoming calls to my cell phone when I'm out and
about - frees up the line and reduces attenuation caused by
an analog trombone. It'd be handy to be able to use it to
transfer terminated calls as well.
Of course if you where on a PRI link, you could do hairpinning,
ect or tromboning and get the call taken back by the PSTN and
transferred to the new number.
--
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 266.4.0 - Release Date: 22/02/2005
___
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: [EMAIL PROTECTED] wrote: Hello
Jim, thx for the answer.. Im happy I found someone that is using
flash :) It's not perfect, but it can be useful. Am I right, if I
transfer a call with flash, the line will be free afterwards ? [...]
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
[Asterisk-Users] Digium E1/T1 card with mgetty+sendfax
Hi, For the project I've used the Eicon DIVA card. It has 8 BRI ports, and for about 25% of the time there are 7 or 8 in use. So we want to replace it with an E1 card. Only issue is, replace it with what? The idea we have been playing with was to get a Digium E1 card (we already have bought lot of Quad E1 cards :-) and then just put it back to back against Asterisk server. And instead of letting mgetty+sendfax talk to /dev/ttyI[0-7], we use /dev/zap/[0-30]. Has anybody else ever tried this? Success- and horror stories are welcome! Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ 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] Digium Card Problems
On Mon, 2005-02-28 at 09:58 +0200, Mark Kidd wrote: Hi all i need urgent help our entire switchboard is down only 5 days after it came up. this is the second time this has happened and i am thinking that asterisk is not worth the trouble it gives. Or you don't know enough about asterisk to be the person people will point at when things go pear shaped. Remember, calling people (or their hard work) names isn't going to propel them towards helping you! Sure, you are under pressure, but you aren't paying us, so we don't feel any of your pressure. Please do read on though... mostly it runs without hassle. but around 2 weeks ago during the test phase we rebooted the machine and did the normal modprobes and this error popped up. coming back to work after the weekend the machine was put on and same thing again please see below. Why would you shut it down just because there is a weekend? Asterisk and linux are designed to be running 24/7 I've always found that you are most likely to experience a problem while re-booting. Once running, things tend to work fine. modprobe zaptel - no problems [EMAIL PROTECTED] root]# modprobe wcfxo /lib/modules/2.4.20-8/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including inva lid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-8/misc/wcfxo.o: insmod /lib/modules/2.4.20-8/misc/wcfxo.o fa iled /lib/modules/2.4.20-8/misc/wcfxo.o: insmod wcfxo failed [EMAIL PROTECTED] root]# modprobe wcfxs /lib/modules/2.4.20-8/misc/wctdm.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including inva lid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-8/misc/wctdm.o: insmod /lib/modules/2.4.20-8/misc/wctdm.o fa iled /lib/modules/2.4.20-8/misc/wctdm.o: insmod wcfxs failed we are running the 4 port fxo digium card. so normal the modprobe wcfxs no problems modules load and board comes up after starting asterisk. i i said this is now the second time this has happened cannot remember what we did to fix this the first time if memory is correct it just decide to work after a while when loading the modules. Try power-off, and wait, and then power-on. Remember to remove the power cord from the back. You want to completely re-set the TDM card, and if you leave the power cord in, it will continue to receive some power. Also, you haven't told us what revision/version of the TDM card you have, nor what version of zaptel you are using. I would suggest you upgrade zaptel asterisk to v1.0.6 if the above doesn't solve the problem. i am now looking like an idiot as i punted this solution rather hard and now we have no board up. You should also have 'punted' an asterisk knowledgeable person. While you may have known enough to get the system this far (and kudos to you fo acheiving that much) you should have a local asterisk 'guru' for times like this. I would suggest you find one soon. Even with a 'standard' PBX solution, you would still need someone to call upon once your limited knowledge of the product isn't enough to solve the problem. please if anyone can help. Next time you might also consider contacting digium. This issue is clearly related to the digium purchased hardware, it is happening before you even try to start asterisk. PS, You might also check the card is properly plugged into the PCI slot, that it does have the extra molex power connector correctly plugged in, etc... Also check the output of lspci, if it doesn't show up there, then it is most likely a hardware problem. Also can try moving it to a different PCI slot (which just forces you to confirm all of my above suggestions :) Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] Possibility of getting someone to delete a user from the list???
On Sun, 2005-02-27 at 17:54 -0500, Robert Webb wrote: This is getting VERY annoying. Is there anyone in here that has access to the list administration to delete the user below??? Pray tell me why. The list isn't being flooded by these messages as far as I see. -- Dave Cotton [EMAIL PROTECTED] ___ 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] How does the g.729 registration program work?
On Mon, Feb 28, 2005 at 12:35:29AM -0330, Paul Fielding wrote: You misunderstand. Ofcourse I need to run the register program on the machine itself. The point is I build them from images and every now and then I roll out a new image. My question is, what do I need to preserve from the previous image to keep the licences. Obviously reformatting the disk and reregistering is not going to work. I could be mistaken, but doesn't the license tie itself to the nics on the server? I believe the Digium server will allow you to reregister as much as you want as long as it's still got the same nics... It does, but if you're only upgrading asterisk but not changing any hardware, there's no need to reregister anything. In my case I just wanted to make sure that the registration wasn't going to write somewhere read-only (say /usr/lib) or in a ram-disk (in my case /etc). It's in /var/lib/asterisk which I have as a real disk... Even if you can reregister each time, it easier to remember the actual licence than it is to remember the key and reenter manually each time. -- Martijn van Oosterhout Ecomtel Pty Ltd ___ 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] Digium E1/T1 card with mgetty+sendfax
On Mon, 28 Feb 2005, Edwin Groothuis wrote: For the project I've used the Eicon DIVA card. It has 8 BRI ports, and for about 25% of the time there are 7 or 8 in use. So we want to replace it with an E1 card. Only issue is, replace it with what? The idea we have been playing with was to get a Digium E1 card (we already have bought lot of Quad E1 cards :-) and then just put it back to back against Asterisk server. And instead of letting mgetty+sendfax talk to /dev/ttyI[0-7], we use /dev/zap/[0-30]. sendfax (and mgetty) requires a modem interface. The zaptel interfaces are raw tdm interfaces. SpanDSP could be made to provide a smartmodem interface but no such code exists yet (as far as I know). Peter ___ 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] Two offices connection
I would like connect two offices where one office have 4 PSTN Analog lines and another office without any PSTN. Both the offices will have two separate Asterisk server with TDM400P cards (4 ports FXS FXO). My questions is that how to configure Asterisk to forward the PSTN calls directly to another Asterisk which has the TDM400P card without pressing the extension number. Diagram like following ---PSTN line1 --[Asterisk]__WAN__[Asterisk ]-Phone Set1 ---PSTN line x -[TDM400P] [TDM400P] Phone Set1 So, call coming from PSTN should go directly to Phone Set1 without any Extension. Is it possible, if so,please let me know how to configure both Asterisk server? Thanking you, Azhar -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. Grameen CyberNet Ltd. ___ 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] call from two sip phones registered in different asterisk server
Hi all Ihave registered my phone1 in asterisk server 192.168.0.9 and phone2 in asterisk server 192.168.0.6. Both are sip phones. i configured the extensions.conf file in both the server. the extensions.conf file on server 192.168.0.9 is exten=301,1,Dial(SIP/[EMAIL PROTECTED],20,tr) exten=401,1,Dial(SIP/phone1,20,tr) 301 is the extension number for phone 2 in asterisk server 192.168.0.6 and 401 is the extension numberf for phone 1in asterisk server 192.168.0.9. Similarly i modified the extension file in asterisk server 192.168.0.6. now i try to make call from phone1 to phone2. i dialed the number 301 and the phone2 is ringing.but when i tried to pick up the phone it's disconnected giving me message call forbidden and terminated. Can anybody help me to show when i could have made mistake in configuring the above configuration. Thanking you all. rajesh Do you Yahoo!? Yahoo! Sports - Sign up for Fantasy Baseball.___ 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] call from two sip phones registered in different asterisk server
Hi all Ihave registered my phone1 in asterisk server 192.168.0.9 and phone2 in asterisk server 192.168.0.6. Both are sip phones. i configured the extensions.conf file in both the server. the extensions.conf file on server 192.168.0.9 is exten=301,1,Dial(SIP/[EMAIL PROTECTED],20,tr) exten=401,1,Dial(SIP/phone1,20,tr) 301 is the extension number for phone 2 in asterisk server 192.168.0.6 and 401 is the extension numberf for phone 1in asterisk server 192.168.0.9. Similarly i modified the extension file in asterisk server 192.168.0.6. now i try to make call from phone1 to phone2. i dialed the number 301 and the phone2 is ringing.but when i tried to pick up the phone it's disconnected giving me message call forbidden and terminated. Can anybody help me to show where i could have made mistake in configuring the above configuration. Thanking you all. rajesh Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. Learn more.___ 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] Two offices connection
On Mon, 2005-02-28 at 20:38, Azhar Chowdhury wrote: I would like connect two offices where one office have 4 PSTN Analog lines and another office without any PSTN. Both the offices will have two separate Asterisk server with TDM400P cards (4 ports FXS FXO). My questions is that how to configure Asterisk to forward the PSTN calls directly to another Asterisk which has the TDM400P card without pressing the extension number. Use the I'net to connect the two offices and IAX2 Diagram like following ---PSTN line1 --[Asterisk]__WAN__[Asterisk ]-Phone Set1 ---PSTN line x -[TDM400P] [TDM400P] Phone Set1 So, call coming from PSTN should go directly to Phone Set1 without any Extension. Is it possible, if so,please let me know how to configure both Asterisk server? Thanking you, Azhar -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] HELP NEEDED ASTERISK AND MEDIATRIX 1102
CAPS LOCK fUnNy On Sun, 2005-02-27 at 20:25, Roy Sigurd Karlsbakk wrote: HELP NEEDED TURNING OFF THE cAPS lOCK KEY :) On Feb 25, 2005, at 20:07, Edward Banfa wrote: Hello all, Hi I would like to know how to configure a Mediatrix 1102 box to work with my asterisk box. I have analog phones that i would like to connect to my Mediatrix box and then connect the Mediatrix box to my asterisk box. My main problems come from the fact that I have limited experience with usiing the two (asterisk and the mediatrix). I know how to use sip.conf , but I am lost when it comes to mediatrix specific configuration. I have search the archives but i have not gotten any thing specific. I would really appreciate any help that can be rendered to set me in the right path. I am desperate here. Thank you all in advance Edward ___ 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 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] Digium Card Problems
On Mon, Feb 28, 2005 at 09:58:28AM +0200, Mark Kidd wrote: Hi all i need urgent help our entire switchboard is down only 5 days after it came up. Read the other email first, you seem to need to know a little more about linux also. In any case I do have one hint for you: [EMAIL PROTECTED] root]# modprobe wcfxo /lib/modules/2.4.20-8/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including inva lid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-8/misc/wcfxo.o: insmod /lib/modules/2.4.20-8/misc/wcfxo.o fa iled /lib/modules/2.4.20-8/misc/wcfxo.o: insmod wcfxo failed After you load a module and it fails, the error message is generally in the kernel logs. Type dmesg to see that. The lines near the end should be helpful. They are necessary for anyone to diagnose your problem... Have a nice day, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ 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] Two offices connection
Hi Howard, Thanks for quick reply. Although I am searching the mailing and googling, do you have a URL about to setup Asterisk with similar situation? Thanking you, Azhar Chowdhury - Original Message - From: Howard Lowndes [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 28, 2005 4:07 PM Subject: Re: [Asterisk-Users] Two offices connection On Mon, 2005-02-28 at 20:38, Azhar Chowdhury wrote: I would like connect two offices where one office have 4 PSTN Analog lines and another office without any PSTN. Both the offices will have two separate Asterisk server with TDM400P cards (4 ports FXS FXO). My questions is that how to configure Asterisk to forward the PSTN calls directly to another Asterisk which has the TDM400P card without pressing the extension number. Use the I'net to connect the two offices and IAX2 Diagram like following ---PSTN line1 --[Asterisk]__WAN__[Asterisk ]-Phone Set1 ---PSTN line x -[TDM400P] [TDM400P] Phone Set1 So, call coming from PSTN should go directly to Phone Set1 without any Extension. Is it possible, if so,please let me know how to configure both Asterisk server? Thanking you, Azhar -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. Grameen CyberNet Ltd. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. Grameen CyberNet Ltd. ___ 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] T.38 fax summary
On Sun, Feb 27, 2005 at 05:32:49PM -0800, Lee Howard wrote: Quite right. I'm sorry to have misled. What happens is this (as an example scenario): The receiver will, for an example, receive the post-page message. The sender expects a response to this. The receiver, however, is required to wait between 55 and 95 ms before transmitting the response. The sender will likely be looking for the post-page response immediately after transmitting the post-page message. Per spec the sender will only wait about 3 seconds (per-spec between 2550 and 3450 ms) before giving up wating and retransmitting the post-page message (and then re-expecting the response). Thank you. So the 1 second lag I suggested is too much, but the principle is sound. Say we change it to half a second you're well under the limit. The question then becomes, is a fixed half-a-second jitterbuffer good enough to remove all the problematic jitter from the signal. This is a testable assertion (though unfortunatly I don't have the necessary equipment), simulating jitter is possible and hopefully the jitterbuffer itself is tunable. A tunable jitterbuffer sounds like a good idea, anyone actually thinking of implementing it though? Have a nice day, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ 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] X100P with Analogue DDI Trunks
I have * configured with 2 X100P cards (fxs_ks). The lines from the telco are 'analogue both way ddi trunks'. This means that every inbound call contains digits that represent an extension on the PBX. I can make outbound calls from * with no problem however I cannot receive inbound calls on these trunks. Some investigation has shown that when the line is idle there is 50 volts present and zttool shows the X100P state as OK. When a call is presented the voltage drops to 0 and the X100P state changes to RED. I believe that at this point the telco is waiting to receive a ringing or busy tone. Has anyone managed to get this configuration working? Am I using the right interface FXO? Thoughts and ideas please. Mike ___ 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] Re: T.38 fax summary
1) Get a 4-port TDM card and install it into your Asterisk box. Connect the TDM ports to your modem ports. Then forward incoming calls on fax DIDs to those TDM ports. Digium TDM 4 fxs is not really a good choice for a faxing system. I've tested it for a while. You should read old messages here about it. I'm using a linksys pap2-na now, it is working well and it does cost less than a digium tdm. Sipura SPA-2000 is also working for a fax system ___ 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] Digium Card Problems
Hi Mark, On Mon, 28 Feb 2005, Mark Kidd wrote: modprobe zaptel - no problems [EMAIL PROTECTED] root]# modprobe wcfxo I'm just curious, did 'modprobe wcfxo' ever work? I seem to remember that for the TDM400P suite, the module to load was (rather confusingly) 'wcfxs', even though you've got FXO modules on the card. we are running the 4 port fxo digium card. so normal the modprobe wcfxs no problems modules load and board comes up after starting asterisk. That's TDM04B, right? If you don't have the Wildcard X100P (or something of the sort) plugged too then I see no reason to be loading 'wcfxo'. Hope that helps. Regards, Gerald. PS: The module name was later changed from 'wcfxs' to 'wctdm' (to avoid confusion I think. So, if you have no X100P, I think you can safely ignore loading 'wcfxo') ___ 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] Possibility of getting someone to delete a user from the list???
On Mon, Feb 28, 2005 at 10:04:27AM +0100, Dave Cotton wrote: On Sun, 2005-02-27 at 17:54 -0500, Robert Webb wrote: This is getting VERY annoying. Is there anyone in here that has access to the list administration to delete the user below??? Pray tell me why. The list isn't being flooded by these messages as far as I see. Their mailserver is broken in that it sends bounces to the From address (ie the person who sent the email) rather than the Sender (the asterisk mail server). So you only get an message from them when you send something. That is, one email for *every* message you send. There's also a server somewhere sending each message back to me with this attached: --- 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. --- However, the content analysis tells me the score is 0.1 of the necessary 5.0. Unfortunatly it's not helpful enough in determining the email address with the problem. Have a nice day, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ 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] HELP NEEDED! - Asterisk GUI
Hi, Its now up at http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20phpconfig I would be interested in any feedback. Hope it helps. C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julius Kidubuka Sent: 28 February 2005 04:50 To: C. Tomlinson Cc: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI I'll look out for it, thanks! Julius. Julius, I have just setup and installed phpconfig with the help of others on this mailing list. I didn't use CVS checkout as I don't have CVS installed. I am about to document the process for the Wiki which I hope will help :) C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julius Kidubuka Sent: 25 February 2005 14:33 To: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI I am having trouble using cvs, is it possible to use cvsup or any other method available and still get to install, configure and use phpconfig? If so, how do I go about it? Julius. Does this mean I have to download and re-compile my asterisk sources inorder to get that file? And if yes, how do I get the sources with cvs checkout phphconfig? If no, how is it done? No, only do the cvs checkout phpconfig, and put the files in the right directory that's all. Guido Hecken ___ 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 -- Rgds, Julius Kidubuka. My advice to you is get married: if you find a good wife you'll be happy; if not, you'll become a philosopher. ___ 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 -- Rgds, Julius Kidubuka. My advice to you is get married: if you find a good wife you'll be happy; if not, you'll become a philosopher. ___ 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: I'll look out for it, thanks! Julius. Julius, I have just setup and installed phpconfig with the help of others on this mailing list. I didn't use CVS checkout as I don't have CVS installed. I am about to document the process for the Wiki which I hope will help :) C 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] Pb DTMF with Asterisk vs Cirpack Transit, Node
Salut Guy, I have the same problem with a Cirpack (B3G carrier) What I see is that you use sip info to detect DTMF. The problem is that there is no normalisation on the content of the sip info frame for dtmf detection. First, asterisk try to detect the header application/dtmf-relay and you have the header application/dtmf see line 6069 of /channels/chan_sip.c function receive_info Second, asterisk use the protocol define by cisco for dtmf sip info. It means it's looking for the word signal into the body. in you body, you have only the digits corresponding to the dtmf so asterisk can't find the dtmf. You have to modify the code around the line 6075 to extract the information. I think it can resolve your problem. hope it's helps. I have some problem detecting DTMF from GSM phone using B3G services (cirpack node also)... do you have also this kind of problem? Florian Lefeuvre Actelium Corporation. Pau. ___ 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] dialout with PPP on ISDN to an ISP
On Sun, Feb 27, 2005 at 10:32:21PM -0600, Steven Critchfield wrote: On Mon, 2005-02-28 at 00:43 +0100, Ilija Poznic wrote: Hello my name is Ilija Poznic and I have a problem. My configuration is 1. Digium TDM4000P with one FXS. 2. AVM Fritz ISDN adapter (configured with capi). When I connect to my ISP and then start *. Asterisks is registering me to SIP provider iconnect. After that I can call international call trough VoIP. My problem is that I want to dialout to ISP only when I have a international call. I tried with PPPD but I can work it out. Anathor solution is by Steven Steven Critchfield Any reason you can't use a .call file to initiate the call? that also I do not understand how to use to make ISP connection. I'm not sure if you can use a .call file. I am not sure you can use ZapRas to call out on a CAPI channel. I am sure the suggestion I was making in the quoted message was to get a .call file dropped off to use ZapRas to dial the ISP. You will probably be unable to use PPP while asterisk has access to the ISDN adapter. Well with capi it should be possible. ppp via pppcapiplugin and asterisk via chan_capi -- Steven Critchfield [EMAIL PROTECTED] -- Tho/\/\as ___ 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] Digium E1/T1 card with mgetty+sendfax
On Mon, Feb 28, 2005 at 04:33:05AM -0600, [EMAIL PROTECTED] wrote: On Mon, 28 Feb 2005, Edwin Groothuis wrote: For the project I've used the Eicon DIVA card. It has 8 BRI ports, and for about 25% of the time there are 7 or 8 in use. So we want to replace it with an E1 card. Only issue is, replace it with what? The idea we have been playing with was to get a Digium E1 card (we already have bought lot of Quad E1 cards :-) and then just put it back to back against Asterisk server. And instead of letting mgetty+sendfax talk to /dev/ttyI[0-7], we use /dev/zap/[0-30]. sendfax (and mgetty) requires a modem interface. The zaptel interfaces are raw tdm interfaces. SpanDSP could be made to provide a smartmodem interface but no such code exists yet (as far as I know). Aha, that makes sense. On a different note, isdn4linux provides such an interface. Only then I need a way have i4l talking with the zaptel driver. Wish I understood more of this stuff... :-/ Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ 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] Possibility of getting someone to delete a user from the list???
Martijn van Oosterhout wrote: On Mon, Feb 28, 2005 at 10:04:27AM +0100, Dave Cotton wrote: On Sun, 2005-02-27 at 17:54 -0500, Robert Webb wrote: This is getting VERY annoying. Is there anyone in here that has access to the list administration to delete the user below??? Pray tell me why. The list isn't being flooded by these messages as far as I see. Their mailserver is broken in that it sends bounces to the From address (ie the person who sent the email) rather than the Sender (the asterisk mail server). So you only get an message from them when you send something. That is, one email for *every* message you send. There's also a server somewhere sending each message back to me with this attached: --- 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. --- However, the content analysis tells me the score is 0.1 of the necessary 5.0. Unfortunatly it's not helpful enough in determining the email address with the problem. Well I must be lucky cause I don't get any of these types of things from the list in quiet a while. So I must be lucky somehow. David Have a nice day, ___ 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] Transfer a call ? Am I lookingfortheflashcommand ?
BTW: Whats actually that SendDTMF ? thing ? http://www.voip-info.org/wiki-Asterisk+cmd+sendDTMF DTMF definition : http://en.wikipedia.org/wiki/DTMF N.B.: please try to trim your answers, the message is becoming pretty long hth ___ 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] ASTERISKBRASIL.ORG
please, all listas.asteriskbrasil.org mailinglist to reconfigure to new IP addresses sen mail to [EMAIL PROTECTED] regards, Max Rivera ___ 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] where is voice conduits
Oups I shouldn't have left that much voice messages those last weeks ;-) I once got to talk with someone from voiceconduits via AIM, but that's all, no reply to emails and voicemail! Marc ross jones wrote: Does any one know what happened with voice conduits? I have been trying to reach them for nearly three weeks now. Their voice mail boxes are full and writing email to them does not get any returns. Thoughts or sightings are appreciated. -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] IT Service Providerhttp://www.msnetworks.lu 15, route d'Esch Phone: +352 2727 3030 L-4450 Belvaux Fax: +352 2727 3060 --- MS Networks powered service --- http://www.LuxAdmin.com Hosting and housing solutions --- ___ 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] Asterisk 1.0.6
Russell Bryant schrieb: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Greetings Everyone! Version 1.0.6 of Asterisk, zaptel, libpri, and Asterisk-addons has been released. There is also a new tarball for Asterisk-sounds. They are available for download on the digium FTP site: ftp://ftp.asterisk.org/pub/asterisk/ ftp://ftp.asterisk.org/pub/zaptel/ ftp://ftp.asterisk.org/pub/libpri/ ChangeLogs are available with the source as well as on the following web page: http://dev.asteriskdocs.org I had found this in the ChangeLogs: [...] -- chan_sip: [...] -- 'restrictcid' now properly works on MySQL peers. [...] Is there already DB-Support for sip.conf in this release? Or is it relating to ast_data? Regards Bastian ___ 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] Asterisk 1.0.6
Asterisk stable still has the old capability of 'sipfriends' and 'iaxfriends' for putting data into MySQL for peers. This is what the changelog note is referring to. If you need more information on either of the above, feel free to browse the voip-info.org website! Have a great day. - Joshua Colp. - Original Message - From: Bastian Schern [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 28, 2005 8:13 AM Subject: Re: [Asterisk-Users] Asterisk 1.0.6 Russell Bryant schrieb: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Greetings Everyone! Version 1.0.6 of Asterisk, zaptel, libpri, and Asterisk-addons has been released. There is also a new tarball for Asterisk-sounds. They are available for download on the digium FTP site: ftp://ftp.asterisk.org/pub/asterisk/ ftp://ftp.asterisk.org/pub/zaptel/ ftp://ftp.asterisk.org/pub/libpri/ ChangeLogs are available with the source as well as on the following web page: http://dev.asteriskdocs.org I had found this in the ChangeLogs: [...] -- chan_sip: [...] -- 'restrictcid' now properly works on MySQL peers. [...] Is there already DB-Support for sip.conf in this release? Or is it relating to ast_data? Regards Bastian ___ 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] Re: Two offices connection
Azhar Chowdhury [EMAIL PROTECTED] writes: I would like connect two offices where one office have 4 PSTN Analog lines and another office without any PSTN. Both the offices will have two separate Asterisk server with TDM400P cards (4 ports FXS FXO). What I would do is use DUNDi. There's an excellent how-to at http://www.voip-info.org/wiki-DUNDi+Enterprise+Configuration+IAX. -tih -- Don't ascribe to stupidity what can be adequately explained by ignorance. ___ 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] SIP broadband phone addon for asterisk
Hi Is there a add-on for asterisk where I can define a rate plan for outgoing international calls and let my sip users make calls depending on the credit they have. tks Kanishka ___ 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] HELP NEEDED! - Asterisk GUI
Thanks for the great job plus all the others that contributed to this. I'll certainly use it and give you feedback. Hi, Its now up at http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20phpconfig I would be interested in any feedback. Hope it helps. C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julius Kidubuka Sent: 28 February 2005 04:50 To: C. Tomlinson Cc: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI I'll look out for it, thanks! Julius. Julius, I have just setup and installed phpconfig with the help of others on this mailing list. I didn't use CVS checkout as I don't have CVS installed. I am about to document the process for the Wiki which I hope will help :) C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julius Kidubuka Sent: 25 February 2005 14:33 To: [EMAIL PROTECTED] Cc: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] HELP NEEDED! - Asterisk GUI I am having trouble using cvs, is it possible to use cvsup or any other method available and still get to install, configure and use phpconfig? If so, how do I go about it? Julius. Does this mean I have to download and re-compile my asterisk sources inorder to get that file? And if yes, how do I get the sources with cvs checkout phphconfig? If no, how is it done? No, only do the cvs checkout phpconfig, and put the files in the right directory that's all. Guido Hecken ___ 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 -- Rgds, Julius Kidubuka. My advice to you is get married: if you find a good wife you'll be happy; if not, you'll become a philosopher. ___ 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 -- Rgds, Julius Kidubuka. My advice to you is get married: if you find a good wife you'll be happy; if not, you'll become a philosopher. ___ 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: I'll look out for it, thanks! Julius. Julius, I have just setup and installed phpconfig with the help of others on this mailing list. I didn't use CVS checkout as I don't have CVS installed. I am about to document the process for the Wiki which I hope will help :) C Content analysis details: (0.1 points, 5.0 required) pts rule name description -- -- 0.1 FORGED_RCVD_HELO Received: contains a forged HELO -- Rgds, Julius Kidubuka. My advice to you is get married: if you find a good wife you'll be happy; if not, you'll become a philosopher. ___ 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] Problem with call hold
I got a very strange problem with call-hold function. For calls that come in from PSTN and route to a SIP extension. If I put the call on hold, I cannot unhold the call after. The caller would be left with hold music forever. A warning message would be shown on the console usually a few seconds after putting the call on hold: WARNING[17428]: chan_sip.c:686 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 2079 (non-critical Response). The same unhold function works fine for calls between SIP extensions. I have searched through wiki but could not find the answer. If somebody can shred some light on the problem, it will be very much appreciated. I'm running the Asterisk stable version at Dec 21, 2004. Thanks ahead. Joseph Shi ___ 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] SIP video problems
hi I'm trying to make video work over SIP between two softphones I can get audio, but video fails sip debug is here http://karlsbakk.net/videotest.log.gz can someone take a look at it, please? roy ___ 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] dialing application - newbie question
I am thinking about a making a web based directory that dials a number with one click. From an overview picture does the below look like the correct way to go about it: web app sends something like the below call file to asterisk Action: Originate Channel: SIP/1010 Context: demo Exten: 1234 Priority: 1 Callerid: 1212 The main problem is the actual phone must be set to auto answer otherwise SIP/1010 would have to pickup before the call is placed. on Polycom this would be done with the ALERT_INFO = ring-answer with a really short ring time - ALERT_INFO being passed to the phone via SetVar on the above. on CISCO 79xx the only way to do it is setup a new line that autoanswers on the phone and configure each phone to do this manually. - Is this still correct? Thanx for any input. Walt. _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ 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] Digium E1/T1 card with mgetty+sendfax
On Mon, 28 Feb 2005, Edwin Groothuis wrote: On Mon, Feb 28, 2005 at 04:33:05AM -0600, [EMAIL PROTECTED] wrote: sendfax (and mgetty) requires a modem interface. The zaptel interfaces are raw tdm interfaces. SpanDSP could be made to provide a smartmodem interface but no such code exists yet (as far as I know). Aha, that makes sense. On a different note, isdn4linux provides such an interface. Only then I need a way have i4l talking with the zaptel driver. Wish I understood more of this stuff... :-/ True, it does provide a smartmodem character device interface. However, it does not provide any analog modem or fax conversions. It works for the digital isdn connection bearers such as V.110 V.120 etc. Still, that would be nice for zaptel devices as well. Peter ___ 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] SIP video problems
Hi Roy, Did you check the video codec on the EyeBeam side ? I think that * works properly only with basic h263. Btw, to start video you have to push manually the start video button (ok, that sounds silly but it's not that intuitive...). We have tested it with no nat, and it works fine in those conditions. Gregoire. Roy Sigurd Karlsbakk wrote: hi I'm trying to make video work over SIP between two softphones I can get audio, but video fails sip debug is here http://karlsbakk.net/videotest.log.gz can someone take a look at it, please? roy ___ 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] Secure IAX Interasterisk authentication ?
Hi, I wonder if I can securely authenticate two Asterisk servers with IAX connection. I know for RSA authentication for IAX2 channel, but that seems to be meant for peer authentication... Has anyone done RSA (or any other secure way) authentication between two Asterisk servers ? Any example ? Thanks in advance, regards, Rob. ___ 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] New Instalation
Hello. I found out about asterisk a few days ago looking for an alternative voip solution to cisco and lucent (they have very expensive solutions). The question is... the company works with 2 E1 incoming lines that go directly to 50 call center agents and the rest must be redirected to other location via internet. I need to guarantee that the system will work on high demand... (we'll have stong rush times...) Does some one has seen asterisk working with that kind of demand? Can some one recomend a harward configuration for that??? Thanks a lot... -- El Rubio ___ 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] Re: T.38 fax summary
1) Get a 4-port TDM card and install it into your Asterisk box. Connect the TDM ports to your modem ports. Then forward incoming calls on fax DIDs to those TDM ports. Digium TDM 4 fxs is not really a good choice for a faxing system. I've tested it for a while. You should read old messages here about it. Faxes coming in from PSTN DID's going directly to a TDM card is a very good and reliable solution. In this setup the fax call can be directly bridged from the PSTN to your fax machine with no codecs or lossy compression, etc. This is exactly the solution I've been using for two production fax setups, and they have worked under heavy usage without any issues since I put them in (about 3 months). ___ 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] test
sorry test. ___ 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] T.38 fax summary
Lee Howard wrote: On 2005.02.27 11:28 Jon Gabrielson wrote: You wouldn't happen to know how to do this would you? I currently have a box with both hylafax and asterisk installed. asterisk handles the dedicated voice lines over a t100p and hylafax handles the dedicated fax lines over a 4port serial card with external modems. It would be really nice if I could have them share the lines instead of having all the lines dedicated to either one or the other. The problem is that the only way to detect whether it is a fax is for asterisk to answer it first and then there is no way that I know of to send it on to hylafax. Sure there is a way. A couple of ways (at least). 1) Get a 4-port TDM card and install it into your Asterisk box. Connect the TDM ports to your modem ports. Then forward incoming calls on fax DIDs to those TDM ports. Currently this doesn't work. A FAX machine connected to a TDM card fails almost every call. It used to work OK. The TDM driver seems to be buggy right now. 2) Get another T1 port in the Asterisk box and get a T1 fax modem and do the same thing. You probably don't have the fax-demand to justify the hardware expense, though. Regards, Steve ___ 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] T.38 fax summary
Lee Howard wrote: On 2005.02.27 09:30 Martijn van Oosterhout wrote: On Sun, Feb 27, 2005 at 09:10:48AM -0800, Lee Howard wrote: Fax cannot handle a one-second delay. As Steve mentions in the article, per-spec fax has some timings (particularly silence in direction switching) set at 75 ms +/- 20 ms. So if the delay gets much larger than 75 ms, then there's likely to be trouble. Now, some fax machines may tolerate larger delays, but that tolerance is beyond the spec, and thus should not be used as a gauge. Something's not right here. Quite right. I'm sorry to have misled. What happens is this (as an example scenario): The receiver will, for an example, receive the post-page message. The sender expects a response to this. The receiver, however, is required to wait between 55 and 95 ms before transmitting the response. The sender will likely be looking for the post-page response immediately after transmitting the post-page message. Per spec the sender will only wait about 3 seconds (per-spec between 2550 and 3450 ms) before giving up wating and retransmitting the post-page message (and then re-expecting the response). This is also slightly wrong. The gaps in the audio stream are specified as 75+-20ms. The response is specified as occuring a *minimum* of 75ms after the received carrier has ceased. So if there is a steady 1000 ms lag between the sender and the receiver (both ways, meaning we assume that both ends could have the 1-second jitter buffer), what will happen is this: The sender will finish transmitting the post-page message. One second later the receiver finishes getting it. The receiver will introduce its own required pause, and add to that the overhead of any processing required, and then it will return the signal. The sender will not get that signal for another 1000 ms. That means that for the total processing of that to occur the 2550 ms danger-zone time is nearly reached. Add to that buffer-time the latency time, and I'd say that you'd be looking at a signal failure quite certainly. In real-world action, however, the 2550-3450 ms danger-zone time is practically never reached. In normal use that time is often very close to 400 ms. So yes, 75 ms latency is not accurate for a command-response interaction between two fax machines. And, per-spec the response could, in theory, sustain a 1000 ms lag. However, that would far-exceed normal behavior, and I'd be surprised if it would not prove fatal to most fax communications. I think this still allows significant buffering - say 500ms - without causing trouble. Extreme buffer would, however, be troublesome. 500ms, less the rollover time needed for the FEC, should give pretty good jitter buffering. Regards, Steve ___ 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] Fax Failing
Title: Fax Failing Hello All, I am trying to set up faxing using [EMAIL PROTECTED] 0.6. I have followed the instructions to the best of my knowledge. When a fax comes in, the system seems to detect OK but does ot manage to make the fax to pdf to email leap. Here is what I saw in the CLI when I tested. Any help would be appreciated. Thanks! Wiley -- Starting simple switch on 'Zap/2-1' -- Executing GotoIf(Zap/2-1, 1?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf(Zap/2-1, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(Zap/2-1, ) in new stack -- Executing Wait(Zap/2-1, 1) in new stack -- Executing SetVar(Zap/2-1, intype=aa_1) in new stack -- Executing Cut(Zap/2-1, intype=intype|-|1) in new stack -- Executing GotoIf(Zap/2-1, 0?7:9) in new stack -- Goto (from-pstn-reghours,s,9) -- Executing GotoIf(Zap/2-1, 0?10:12) in new stack -- Goto (from-pstn-reghours,s,12) -- Executing Goto(Zap/2-1, aa_1|s|1) in new stack -- Goto (aa_1,s,1) -- Executing DigitTimeout(Zap/2-1, 3) in new stack -- Set Digit Timeout to 3 -- Executing ResponseTimeout(Zap/2-1, 7) in new stack -- Set Response Timeout to 7 -- Executing BackGround(Zap/2-1, custom/aa_1) in new stack -- Playing 'custom/aa_1' (language 'en') -- Redirecting Zap/2-1 to fax extension == Spawn extension (aa_1, fax, 0) exited non-zero on 'Zap/2-1' -- Executing Goto(Zap/2-1, ext-fax|in_fax|1) in new stack -- Goto (ext-fax,in_fax,1) -- Executing GotoIf(Zap/2-1, 1?2:analog_fax|1) in new stack -- Goto (ext-fax,in_fax,2) -- Executing Macro(Zap/2-1, faxreceive) in new stack -- Executing SetVar(Zap/2-1, FAXFILE=/var/spool/asterisk/fax/1109597736.32.tif) in new stack -- Executing SetVar(Zap/2-1, [EMAIL PROTECTED]) in new stack -- Executing RxFAX(Zap/2-1, /var/spool/asterisk/fax/1109597736.32.tif) in new stack -- Executing Hangup(Zap/2-1, ) in new stack == Spawn extension (ext-fax, h, 1) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' -- Hungup 'Zap/1-1' ___ 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
Re: [Asterisk-Users] Digium E1/T1 card with mgetty+sendfax
Peter Svensson wrote: On Mon, 28 Feb 2005, Edwin Groothuis wrote: For the project I've used the Eicon DIVA card. It has 8 BRI ports, and for about 25% of the time there are 7 or 8 in use. So we want to replace it with an E1 card. Only issue is, replace it with what? The idea we have been playing with was to get a Digium E1 card (we already have bought lot of Quad E1 cards :-) and then just put it back to back against Asterisk server. And instead of letting mgetty+sendfax talk to /dev/ttyI[0-7], we use /dev/zap/[0-30]. sendfax (and mgetty) requires a modem interface. The zaptel interfaces are raw tdm interfaces. SpanDSP could be made to provide a smartmodem interface but no such code exists yet (as far as I know). Such code has been sitting in a near complete state for about 3 or 4 month, with no time available to finish it :-( Regards, Steve ___ 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] where is voice conduits
ross jones wrote: Does any one know what happened with voice conduits? I have been trying to reach them for nearly three weeks now. Their voice mail boxes are full and writing email to them does not get any returns. Thoughts or sightings are appreciated. There was a thread a month or two ago on here about voiceconduits. The general gist was they are not yet open for public business. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] phpconfig
Hello, I recently downloaded phpconfig from http://asterisk.espia-net.net/horde/chora/cvs.php/phpconfig?login=2 but on installing it, my interface does not look like the one at http://rd.it.utah.edu/phpconfig/. The main differences are: 1)On opening a file for editing, on the left menu mine has ony two links i.e Header and the filename.conf as opposed to the deffrent sections on the demo site. Even when i click Header, the page just refreshes and doesn't pick out only the header. 2) I also lack the other links on the right which are in most cases numbers Questions: 1) Am i using an older version? If so, where can i get a newr version? 2) Am i missing some configuration, which one? Thanks in advance, Allan ___ 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] dialing application - newbie question
w fm3 wrote: on CISCO 79xx the only way to do it is setup a new line that autoanswers on the phone and configure each phone to do this manually. - Is this still correct? There was a recent post, look at lists.digium.com for the archives, that detailed an 'auto-answer' script for the 79XX phones. It basically telnets into the phone and presses the answer key! It works, it might not be pretty, but it works. You do have to be careful though, as you don't want this script to execute if you're currently doing anything on your phone - including talking to someone. -Chris ___ 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] phpconfig
Questions: 1) Am i using an older version? If so, where can i get a newr version? 2) Am i missing some configuration, which one? See this newly created document, it explains everything you need to make it work. http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20phpconfig It's been written with the help of peoples on this list. hth ___ 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] Unable to handle ROSE operation 34
Hi, i am getting the follwing messages with asterisk 1.0.5 [...] Feb 28 16:13:05 VERBOSE[8899]: !! Unable to handle ROSE operation 34 [...] Can anybody gibe me a hint what is is about ? Greetings, Martin ___ 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] Re: T.38 fax summary
So are you saying that in my setup where I have a adit 600 channel bank with FXO/FXS connected to a t110p, that asterisk does an analog bridge? Presumably that would mean 56k modems, etc.. would also work fine. I was under the impression that asterisk used iax2 for the internal trunk. Jon. On Monday 28 February 2005 08:10 am, Noah Miller wrote: 1) Get a 4-port TDM card and install it into your Asterisk box. Connect the TDM ports to your modem ports. Then forward incoming calls on fax DIDs to those TDM ports. Digium TDM 4 fxs is not really a good choice for a faxing system. I've tested it for a while. You should read old messages here about it. Faxes coming in from PSTN DID's going directly to a TDM card is a very good and reliable solution. In this setup the fax call can be directly bridged from the PSTN to your fax machine with no codecs or lossy compression, etc. This is exactly the solution I've been using for two production fax setups, and they have worked under heavy usage without any issues since I put them in (about 3 months). ___ 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] help
help I just want a list of commands, if this mail shows in the list, sorry, my bad. ___ 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] Zaptel (Junghanns 4BRI card) to cell phoneproblem
On Fri, 2005-02-25 at 16:29 -0700, Mr. James W. Laferriere wrote: Hello Mark , C. All , Is this device available for sale in the US ? All the digging I've only found outside US mentions of sales . Any help appreciated . JimL No idea. The Unit I have is a locally manufactured device called Digi-Cell - frmaritz (at) global.co.za is the email address on the box it came in Its probably 900Mhz GSM only - in the US - You'll need a 1900Mhz unit??? On Fri, 25 Feb 2005, Mark Elkins wrote: On Fri, 2005-02-25 at 13:46 +, C. Tomlinson wrote: Did you have to make any changes to use the premicell, or was it as simple as an outgoing landline call? I am looking into doing this as you can get deals where calls between chosen numbers are free :-) Absolutely no changes at all I did stick a Phone onto the 2-wire input of the 'PremiCell' to check that all worked - before going via Asterisk - but thats all. [part of the previous message] In South Africa, I have a 4-port ISDN BRI (Euro-ISDN) with bristuff. Calls to Cell phones are no different to any other call... I also added a Digium 4-port analogue card - and have a 'PremiCell' connected to a Trunk line. The PremiCell is a fixed cell device that gives dial-tone in the same way that a Telcom Trunk line would work - except there is no copper to he exchange - just a stubby cellphone antenna. In South Africa it is MUCH MUCH cheaper to make a Cell to Cell call than from Telcom to Cell I'm surprised that more people do not put down a 'PremiCell' type device and route all Cell calls out through it... -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ 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] Fax Failing
looks good. did you install the tif to pdf stuff? (type help-aah for help on how to do this) --- Wiley Siler [EMAIL PROTECTED] wrote: Hello All, I am trying to set up faxing using [EMAIL PROTECTED] 0.6. I have followed the instructions to the best of my knowledge. When a fax comes in, the system seems to detect OK but does ot manage to make the fax to pdf to email leap. Here is what I saw in the CLI when I tested. Any help would be appreciated. Thanks! Wiley -- Starting simple switch on 'Zap/2-1' -- Executing GotoIf(Zap/2-1, 1?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf(Zap/2-1, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(Zap/2-1, ) in new stack -- Executing Wait(Zap/2-1, 1) in new stack -- Executing SetVar(Zap/2-1, intype=aa_1) in new stack -- Executing Cut(Zap/2-1, intype=intype|-|1) in new stack -- Executing GotoIf(Zap/2-1, 0?7:9) in new stack -- Goto (from-pstn-reghours,s,9) -- Executing GotoIf(Zap/2-1, 0?10:12) in new stack -- Goto (from-pstn-reghours,s,12) -- Executing Goto(Zap/2-1, aa_1|s|1) in new stack -- Goto (aa_1,s,1) -- Executing DigitTimeout(Zap/2-1, 3) in new stack -- Set Digit Timeout to 3 -- Executing ResponseTimeout(Zap/2-1, 7) in new stack -- Set Response Timeout to 7 -- Executing BackGround(Zap/2-1, custom/aa_1) in new stack -- Playing 'custom/aa_1' (language 'en') -- Redirecting Zap/2-1 to fax extension == Spawn extension (aa_1, fax, 0) exited non-zero on 'Zap/2-1' -- Executing Goto(Zap/2-1, ext-fax|in_fax|1) in new stack -- Goto (ext-fax,in_fax,1) -- Executing GotoIf(Zap/2-1, 1?2:analog_fax|1) in new stack -- Goto (ext-fax,in_fax,2) -- Executing Macro(Zap/2-1, faxreceive) in new stack -- Executing SetVar(Zap/2-1, FAXFILE=/var/spool/asterisk/fax/1109597736.32.tif) in new stack -- Executing SetVar(Zap/2-1, [EMAIL PROTECTED]) in new stack -- Executing RxFAX(Zap/2-1, /var/spool/asterisk/fax/1109597736.32.tif) in new stack -- Executing Hangup(Zap/2-1, ) in new stack == Spawn extension (ext-fax, h, 1) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' -- Hungup 'Zap/1-1' ___ 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 __ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail ___ 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] phpconfig
Hello, That's the document i read and got all the relevant links. I also tried to follow all the predures . More help is appreciated, Thanks very much Allan On Mon, 28 Feb 2005 10:13:02 -0500, Time Bandit [EMAIL PROTECTED] wrote: Questions: 1) Am i using an older version? If so, where can i get a newr version? 2) Am i missing some configuration, which one? See this newly created document, it explains everything you need to make it work. http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20phpconfig It's been written with the help of peoples on this list. hth ___ 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] Anybody using X-Lite Softphone ? tryed toforwarda call to X-Lite....
Try the snom soft phone! http://snom.com CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Chase Sent: Saturday, February 26, 2005 12:31 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Anybody using X-Lite Softphone ? tryed toforwarda call to X-Lite XLite does not support transfer... You have to buy their XPro -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mateo Meier Sent: Tuesday, February 22, 2005 3:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Anybody using X-Lite Softphone ? tryed to forwarda call to X-Lite Hey Guys Im trying to forward a call from the asterisk mainmenue to my second computer with X-Lite installed.. What I've done so far is this: Installed X-lite @my win PC.. X-Lite configuration: Menu | System Settings | SIP Proxy | default Display Name: mateo01 User Name Authorization User: mateo01 Password: Domain/Realm: 192.168.1.** SIP Proxy: 192.168.1.** 192.168.1.** = IP address of Asterisk and the sip.conf file looks like that: [mateo01] type=friend username=mateo01 callerid=mateo01 1234 host=dynamic secret= disallow=all allow=gsm allow=ulaw allow=alaw context=sip nat=no Now, Im unsure what to do ? whats next ? and what do I type in to extensions.conf instead of the following: exten=2,1,Dial(capi/720:078***) Thx for the help Mateo ___ 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 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] Zaptel (Junghanns 4BRI card) to cell phoneproblem
http://www.psitek.co.za/gsm.html These guys are also in RSA, and Australia. This unit does exactly the same as the DigiCell, which mark is talking about, but is a much better product (and more expensive) maybe they export ? -Herman On Mon, 2005-02-28 at 17:21, Mark Elkins wrote: On Fri, 2005-02-25 at 16:29 -0700, Mr. James W. Laferriere wrote: Hello Mark , C. All , Is this device available for sale in the US ? All the digging I've only found outside US mentions of sales . Any help appreciated . JimL No idea. The Unit I have is a locally manufactured device called Digi-Cell - frmaritz (at) global.co.za is the email address on the box it came in Its probably 900Mhz GSM only - in the US - You'll need a 1900Mhz unit??? On Fri, 25 Feb 2005, Mark Elkins wrote: On Fri, 2005-02-25 at 13:46 +, C. Tomlinson wrote: Did you have to make any changes to use the premicell, or was it as simple as an outgoing landline call? I am looking into doing this as you can get deals where calls between chosen numbers are free :-) Absolutely no changes at all I did stick a Phone onto the 2-wire input of the 'PremiCell' to check that all worked - before going via Asterisk - but thats all. [part of the previous message] In South Africa, I have a 4-port ISDN BRI (Euro-ISDN) with bristuff. Calls to Cell phones are no different to any other call... I also added a Digium 4-port analogue card - and have a 'PremiCell' connected to a Trunk line. The PremiCell is a fixed cell device that gives dial-tone in the same way that a Telcom Trunk line would work - except there is no copper to he exchange - just a stubby cellphone antenna. In South Africa it is MUCH MUCH cheaper to make a Cell to Cell call than from Telcom to Cell I'm surprised that more people do not put down a 'PremiCell' type device and route all Cell calls out through it... ___ 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] Sipura SPA-841 autodial?
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
RE: [Asterisk-Users] Fax Failing
I thought so. I ran install-pdf from the command line after installing everything else. Did I miss something? Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, February 28, 2005 8:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Fax Failing looks good. did you install the tif to pdf stuff? (type help-aah for help on how to do this) --- Wiley Siler [EMAIL PROTECTED] wrote: Hello All, I am trying to set up faxing using [EMAIL PROTECTED] 0.6. I have followed the instructions to the best of my knowledge. When a fax comes in, the system seems to detect OK but does ot manage to make the fax to pdf to email leap. Here is what I saw in the CLI when I tested. Any help would be appreciated. Thanks! Wiley -- Starting simple switch on 'Zap/2-1' -- Executing GotoIf(Zap/2-1, 1?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf(Zap/2-1, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(Zap/2-1, ) in new stack -- Executing Wait(Zap/2-1, 1) in new stack -- Executing SetVar(Zap/2-1, intype=aa_1) in new stack -- Executing Cut(Zap/2-1, intype=intype|-|1) in new stack -- Executing GotoIf(Zap/2-1, 0?7:9) in new stack -- Goto (from-pstn-reghours,s,9) -- Executing GotoIf(Zap/2-1, 0?10:12) in new stack -- Goto (from-pstn-reghours,s,12) -- Executing Goto(Zap/2-1, aa_1|s|1) in new stack -- Goto (aa_1,s,1) -- Executing DigitTimeout(Zap/2-1, 3) in new stack -- Set Digit Timeout to 3 -- Executing ResponseTimeout(Zap/2-1, 7) in new stack -- Set Response Timeout to 7 -- Executing BackGround(Zap/2-1, custom/aa_1) in new stack -- Playing 'custom/aa_1' (language 'en') -- Redirecting Zap/2-1 to fax extension == Spawn extension (aa_1, fax, 0) exited non-zero on 'Zap/2-1' -- Executing Goto(Zap/2-1, ext-fax|in_fax|1) in new stack -- Goto (ext-fax,in_fax,1) -- Executing GotoIf(Zap/2-1, 1?2:analog_fax|1) in new stack -- Goto (ext-fax,in_fax,2) -- Executing Macro(Zap/2-1, faxreceive) in new stack -- Executing SetVar(Zap/2-1, FAXFILE=/var/spool/asterisk/fax/1109597736.32.tif) in new stack -- Executing SetVar(Zap/2-1, [EMAIL PROTECTED]) in new stack -- Executing RxFAX(Zap/2-1, /var/spool/asterisk/fax/1109597736.32.tif) in new stack -- Executing Hangup(Zap/2-1, ) in new stack == Spawn extension (ext-fax, h, 1) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' -- Hungup 'Zap/1-1' ___ 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 __ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail ___ 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] Anybody using X-Lite Softphone ? tryedtoforwarda call to X-Lite....
Mateo, Dialing the extension to your softphone is the same as any hardware extension. Exten = 1000,1,Dial,(SIP/1000,20,trf) pretty exten = 1000,2,Macro(vmessage,1000) exten = 1000,3,Hangup Change [mateo01] to [1000] in your sip and you will be saying that ext. 1000 is registered with the specifics you are using. Update the settings in your softphone to register the name and number as 1000 Then any attempt to dial 1000 should come to that phone. Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Stredicke Sent: Monday, February 28, 2005 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Anybody using X-Lite Softphone ? tryedtoforwarda call to X-Lite Try the snom soft phone! http://snom.com CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Chase Sent: Saturday, February 26, 2005 12:31 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Anybody using X-Lite Softphone ? tryed toforwarda call to X-Lite XLite does not support transfer... You have to buy their XPro -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mateo Meier Sent: Tuesday, February 22, 2005 3:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Anybody using X-Lite Softphone ? tryed to forwarda call to X-Lite Hey Guys Im trying to forward a call from the asterisk mainmenue to my second computer with X-Lite installed.. What I've done so far is this: Installed X-lite @my win PC.. X-Lite configuration: Menu | System Settings | SIP Proxy | default Display Name: mateo01 User Name Authorization User: mateo01 Password: Domain/Realm: 192.168.1.** SIP Proxy: 192.168.1.** 192.168.1.** = IP address of Asterisk and the sip.conf file looks like that: [mateo01] type=friend username=mateo01 callerid=mateo01 1234 host=dynamic secret= disallow=all allow=gsm allow=ulaw allow=alaw context=sip nat=no Now, Im unsure what to do ? whats next ? and what do I type in to extensions.conf instead of the following: exten=2,1,Dial(capi/720:078***) Thx for the help Mateo ___ 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 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] Zaptel (Junghanns 4BRI card) to cell phoneproblem
On Mon, 2005-02-28 at 17:50 +0200, Herman Cremer wrote: http://www.psitek.co.za/gsm.html These guys are also in RSA, and Australia. This unit does exactly the same as the DigiCell, which mark is talking about, but is a much better product (and more expensive) maybe they export ? Except that when I've been to the USA - I've needed a 1900Mhz phone - this is only 900 and 1800... *GSM INTERFACE * GSM output 900MHz: Class 4/5, 2W EGSM * GSM output 1800MHz: Class 1, 1W DCS * SIM interface: 3V mini SIM but look at the website (Hey, it looks like my box!) as the features are what you are looking for... I believe Motorola was one of the earlier producers of this type of device - but would think that most of the manufacturers would have a similar type of unit. Push the Telephone access in disaster areas, where wire-network infrastructure is damaged point... :-) -Herman On Mon, 2005-02-28 at 17:21, Mark Elkins wrote: On Fri, 2005-02-25 at 16:29 -0700, Mr. James W. Laferriere wrote: Hello Mark , C. All , Is this device available for sale in the US ? All the digging I've only found outside US mentions of sales . Any help appreciated . JimL No idea. The Unit I have is a locally manufactured device called Digi-Cell - frmaritz (at) global.co.za is the email address on the box it came in Its probably 900Mhz GSM only - in the US - You'll need a 1900Mhz unit??? On Fri, 25 Feb 2005, Mark Elkins wrote: On Fri, 2005-02-25 at 13:46 +, C. Tomlinson wrote: Did you have to make any changes to use the premicell, or was it as simple as an outgoing landline call? I am looking into doing this as you can get deals where calls between chosen numbers are free :-) Absolutely no changes at all I did stick a Phone onto the 2-wire input of the 'PremiCell' to check that all worked - before going via Asterisk - but thats all. [part of the previous message] In South Africa, I have a 4-port ISDN BRI (Euro-ISDN) with bristuff. Calls to Cell phones are no different to any other call... I also added a Digium 4-port analogue card - and have a 'PremiCell' connected to a Trunk line. The PremiCell is a fixed cell device that gives dial-tone in the same way that a Telcom Trunk line would work - except there is no copper to he exchange - just a stubby cellphone antenna. In South Africa it is MUCH MUCH cheaper to make a Cell to Cell call than from Telcom to Cell I'm surprised that more people do not put down a 'PremiCell' type device and route all Cell calls out through it... ___ 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: http://www.psitek.co.za/gsm.html These guys are also in RSA, and Australia. This unit does exactly the same as the DigiCell, which mark is talking about, but is a much better product (and more expensive) [...] Content analysis details: (0.9 points, 5.0 required) pts rule name description -- -- 0.1 FORGED_RCVD_HELO Received: contains a forged HELO 0.8 CELL_PHONE_IMPROVE BODY: Talks about cell-phone signal improvement -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ 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] Dial out through Broadvoice
On Saturday February 26 2005 4:45 pm, John Millican wrote:
On Saturday February 26 2005 4:30 pm, Chris Ford wrote:
I tried to call you number to see what I would get and you have a verizon
Voice messaging service.
if you called the 6037862111 that is a voicemail number tyhat i was calling
to test knowing it would not be busy and would not bother anyone.
Make sure you have your iax set up right in the Iax.conf and your
outbaound registering string going back out.
I have mine set up that I dial 6 to get out on my broadvoice line and 9
to get out on my voice pulse line.
I am not using IAX at all. Did not think broadvoice supported it, am I
wrong?
More Comments at BOTTOM
Hello all,
When I call the Broadvoice number all is good.
When I try to call out through DISA on my broadvoice line i get the
following:
Executing Dial(SIP/147.135.0.129-0815bc60,
SIP/[EMAIL PROTECTED]|30) in new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 480 Temporarily Not Available back from
147.135.16.128
-- SIP/proxy.bos.broadvoice.com-3493 is circuit-busy
== Everyone is busy/congested at this time
-- Executing Busy(SIP/147.135.0.129-0815bc60, ) in new stack
== Spawn extension (outgoing, 16037862111, 102) exited non-zero on
'SIP/147.135.0.129-0815bc60'
Is this as simple as it seems? Broadvoice is circut busy? Can any
one think
of any other reason I might get this message? Or do I just need to
call
BroadVoice and complain? I have tried two different proxy's (ip's in
/etc/hosts) and get the same error.
in extensions.conf:
[outgoing]
exten = _1NXXNXX, 1, dial(SIP/${EXTEN}
@proxy.bos.broadvoice.com,30) ;
exten = _1NXXNXX, 2, congestion() ; No answer, nothing
exten = _1NXXNXX, 102, busy() ; Busy
in sip.conf:
[general]
context=default ; Default context for incoming
calls
port=5060 ; UDP Port to bind to (SIP standard
port is 5060)
bindaddr=192.168.123.100 ; IP address to bind to
(0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound
calls
; Note: Asterisk only uses the first
host
; in SRV records
; Disabling DNS SRV lookups disables
the
; ability to place SIP calls based on
domain
; names to some other SIP users on the
Internet
register =
[EMAIL PROTECTED]:XX:[EMAIL PROTECTED]
[broadvoice1]
type=friend
username=603XXX
fromuser=603XXX
secret=XX
host=proxy.bos.broadvoice.com
fromdomain=sip.broadvoice.com
context=broadvoice
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no
nat=yes
[bv-in-1]
type=friend
host=sip.broadvoice.com
context=broadvoice
dtmfmode=inband
canreinvite=no
nat=yes
Try adding this line to sip:
insecure=very
Added insecure=very and same message
see if that helps. if not, try a standard registration string instead
of the one broadvoice tells you to use.
Also - make sure you're using the password they sent you in an email -
not the one you used when you signed up on their website.
Registration seems to work and shows as registered when i run sip show
registry.
I have been on the support line with broadvoice several times now and still no
resolution, so I am askking help again, please. Below is sip show registry
and a sip debug. Does any one have any sugestions? I followed the
instrution at http://edvina.net/broadvoice/ along with others and sytill no
luck on outbound calls. in the sip debug it is showing the internal ip in
the callid field. I have externip= external ip in sip .conf am i missing
something else?
linux*CLI sip show registry
HostUsername Refresh State
sip.broadvoice.com:5060 [EMAIL PROTECTED]15 Registered
linux*CLI sip debug
SIP Debugging Enabled
Feb 28 11:08:34 NOTICE[5214]: chan_sip.c:3999 sip_reregister:--
Re-registration for [EMAIL PROTECTED]@sip.broadvoice.com
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.123.100:5060;branch=z9hG4bK01ee8f5f
From: sip:[EMAIL PROTECTED];tag=as2999a955
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: sip:[EMAIL PROTECTED]
Event: registration
Content-Length: 0
(no NAT) to 147.135.8.128:5060
linux*CLI
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.123.100:5060;received=69.160.185.49;branch=z9hG4bK01ee8f5f;rport=63364
From: sip:[EMAIL PROTECTED];tag=as2999a955
To: sip:[EMAIL PROTECTED];tag=SD30scc99-
Call-ID: [EMAIL PROTECTED]
CSeq: 104 REGISTER
Contact: sip:[EMAIL PROTECTED];expires=20
Content-Length: 0
8 headers, 0 lines
Feb 28 11:08:34 NOTICE[5214]: chan_sip.c:6779 handle_response: Outbound
Registration: Expiry for sip.broadvoice.com is 20 sec
Re: [Asterisk-Users] Re: T.38 fax summary
On Mon, 2005-02-28 at 09:15 -0600, Jon Gabrielson wrote: So are you saying that in my setup where I have a adit 600 channel bank with FXO/FXS connected to a t110p, that asterisk does an analog bridge? Presumably that would mean 56k modems, etc.. would also work fine. I was under the impression that asterisk used iax2 for the internal trunk. Even if it used IAX2 internally, it wouldn't degrade quality, but internally it doesn't have to incur the overhead of the IP network as it doesn't go anywhere so it doesn't use IAX2. 56k modems will only work if you only are analog for the first hop and then you are digital(T1) to the PSTN then. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] queue_log and exitwithkey
Hello, I am using Asterisk stable and have a question about the queue_log. It seems like in the past (although I can't find my old logs) that the exitwithkey produced a wait time entry. It would seem logical that you would want to track this. Right now it only shows the key they pressed, and the position they were in. I want to know how long they waited before they bailed. Right now I am circumventing this by having the keypress call an AGI that determines this by the epoch and sending it to the log, but it seems like it is much better suited for inside app_queue. Is this by design? Would anyone want this as a feature? It seems like an easy thing to do, I'm just not up to the challenge of doing it. I'm sure that I could find someone who would though. Any ideas? Thanks, Chuji ___ 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] FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format
Hello, I've got problems to install zaptel on a SuSE 9.1 System. The System has got a Linux 2.6.9 Kernel. If I try to load zaptel framework (modprobe zaptel) I get this message: FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format How I can fix this. At compile time, there were no Errors. Regards Bastian ___ 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] Weird behaviour on incoming DIDs
Folks,
I have a problem here. I have 2 DIDs, one a 415 number
and the other a 650 number. I have my extensions.conf
set up to handle both of them exactly the same way,
passing them to an internal context. When _I_ dial
either DID, I get exactly the same behaviour that I
have specified (the call is answered, and then I play
my own welcome mesage, then handle any extension
dialed).
However, when one of my friends dials in, the 415 DID
consistently works as designed, but the 650 DID
sometimes just tells him goodbye, and then hangs up on
him!
I don't know if it matters, but I am calling from the
650 area code and he's calling sometimes from the 415
area code and sometimes from 408. No, there is no
pattern as to which incoming call gets hung-up!
Here are the relevant sections of my extensions.conf:
[incoming_context]
; This is the incoming call/DID context only.
exten = 415xxx,1,Goto(internal_context,s,1)
exten = 650xxx,1,Goto(internal_context,s,1)
; munged numbers, obviously
exten = i,1,Background(invalid)
exten = #,1,Background(goodbye)
exten = #,2,Wait(2)
exten = #,3,Hangup
exten = t,1,Background(goodbye)
exten = t,2,Wait(2)
exten = t,3,Hangup
exten = h,1,Hangup
exten = 1000,1,Background(goodbye);
exten = 1000,2,Wait(2);
exten = 1000,3,Hangup
#include other_extensions.conf
And, my other_extensions.conf has:
[internal_context]
exten = #,1,Goto(incoming_context,1000,1)
exten = *,1,VoiceMailMain()
exten = *,2,Background(demo-congrats)
exten = h,1,Goto(incoming_context,h,1))
exten = i,1,Background(invalid)
exten = s,1,Answer()
exten = s,2,Background(test-welcome)
exten = t,1,Goto(incoming_context,1000,1)
exten = _[1-9]XX,1,VoiceMail(u${EXTEN})
exten = _[1-9]XX,2,Goto(incoming_context,1000,1)
Also, when this happens, I don't see Asterisk logging
the hung-up call in Master.csv. Other calls seem to be
logged fine.
Can this by any chance be caused by Master.csv getting
too large? If so, how come the same asterisk can still
handle calls coming in on the other DID with no
problems?
Thanks,
Maya
~
__
Do you Yahoo!?
Take Yahoo! Mail with you! Get it on your mobile phone.
http://mobile.yahoo.com/maildemo
___
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] Passing additional information to an AGI via a call file
I have a desire to incorporate asterisk into some of my network monitoring. I would like to use the outgoing calling features to connect a phone (on-call cell phones) to an agi script which can provide some information to the called party. Ultimately I would like to pass 2 pieces of information to the agi, first an integer that is representative of a problem, and second a unix timestamp of the time the problem occurecd, so that my AGI could string together the phrases Error 17 Occured on Monday Feb 28 2005 at 10:59 a.m. The actual agi is pretty much done I just need to figure out how to pass the two new pieces of information to it from the .call file. Any suggestions, or resrources I should be reading? ___ 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] IAX2 (Stupid question)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of leandro_tenorio Sent: Sunday, February 27, 2005 8:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] IAX2 (Stupid question) at least 4 me. Anyone knows what are the variables in an inbound IAX2 call who reflect the actual codec and DNID, DNIS, original peer description, I'm only able to see it during an iax debug Timestamp: 3ms SCall: 1 DCall: 0 [66.98.146.34:5036] VERSION : 2 CALLED NUMBER : 911214686 CALLING NUMBER : asterisk CALLING NAME: asterisk LANGUAGE: en USERNAME: tenorio FORMAT : 2 CAPABILITY : 18 ADSICPE : 2 DATE TIME : 173807980 Tkx. ___ 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] x101p + Nortel ATA2
Hi, Does anyone have any experience connecting Asterisk to a Meridian system using an ATA2 and x101p? The basics work -- I can make outbound calls, receive inbound, and use flash to transfer calls, but certain things do not work, specifically with calls from internal extensions. - Does the Meridian/ATA2 pass any kind of callerid info? We do not have external callerid, but I'm not even getting extension numbers. - Calls involving an external line can use DTMF, but calls from another internal extension cannot. This is a problem for voicemail! I've tried *809 but it hasn't helped. Is this a limitation of the Meridian which won't pass DTMF internally? - Calls from internal extensions do not detect hangup properly, external calls are ok. So if an internal extension calls to leave a voicemail, the recording goes on until I do a sofft-hangup from the CLI. The plan was to use Asterisk as a voicemail server, but those three issues make this setup completely useless for that! Thanks, -Gary ___ 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] Suse 9.2 + CAPI Driver
Hello, I'm trying to install CAPI Driver for Suse 9.2 and I found the documentation for this pretty old since It refers toSuse 8.2 ( http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install).This is especially apparent when I look at the section of these instructions for altering "src.drv/makefile" toreplace the occurance of "CARD_PATH". I tried to install the driverusing the default makefile, with the final result of "capi not installed - No such device or address (6)" Are there any updated documents out there? Regards, Victor. ___ 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] Anybody using X-Lite Softphone ? tryedtoforwarda call to X-Lite....
Change [mateo01] to [1000] in your sip and you will be saying that ext.
1000 is registered with the specifics you are using.
Update the settings in your softphone to register the name and number as
1000
Then any attempt to dial 1000 should come to that phone.
Wiley
After doing thoses changes, you can also simplify your dialplan with
something like this:
exten = _1XXX,1,Dial(SIP/${EXTEN},20,tr)
exten = _1XXX,2,Voicemail(u${EXTEN})
exten = _1XXX,3,Hangup
exten = _1XXX,102,Voicemail(b${EXTEN})
exten = _1XXX,103,Hangup
This will match any extension dialed in the 1000-1999 range so you
don't have to write new rules each time you add a phone
hth
___
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 to limit a peer to one connection only?
How can I make sure that I only connect to a peer once? E.g., I want that all my staff only use one sipgate connection to dial out (although I am sure sipgate would love to make more money and let all my staff call out)? bye Ronald ___ 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] making ASTCC web page secure ???
[EMAIL PROTECTED] wrote: How do you make the page http://hostname/cgi-bin/astcc-admin/astcc-admin.cgi 1. use a virtual domain for it, which you do not broadcast 2. use a different name for astcc-admin/ 3. limit it to known IP address from where you log in 4. use .httaccess or httpd.conf Directory limitation, ... bye Ronald secure ? , so that only the person administering the calling cards can see the page and make changes to the calling cards, I was thinking of using .htaccess to restrict the access to the page by requiring a password, however since it is a cgi script that does not seem to be posible. Any ideas, any suggestions ? ___ 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 -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ 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] FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format
Bastian Schern wrote: Hello, I've got problems to install zaptel on a SuSE 9.1 System. The System has got a Linux 2.6.9 Kernel. If I try to load zaptel framework (modprobe zaptel) I get this message: FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format How I can fix this. At compile time, there were no Errors. Regards Bastian Did you do a make linux26 in the zaptel directory? -- Kristian Kielhofner ___ 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] Zaptel (Junghanns 4BRI card) to cell phoneproblem
There are a number of models similar to this, they generally go under the name of 'fixed cellular terminals.' Most of the gsm cell manufacturers make them... for example, nokia makes the noia 22 and 32 models (the 22 is hard to get now) Eurotech have some cheap models using wavecom gsm modules, badged under the winner range, and Burnside make nice units using Siemens TC35 modules. I'm located in UK and IReland currently and can source these locally to me, however I'd guess that delivery to anywhere outside europe would be expensive. Mike Preston [EMAIL PROTECTED] On Mon, 28 Feb 2005 18:07:42 +0200, Mark Elkins [EMAIL PROTECTED] wrote: On Mon, 2005-02-28 at 17:50 +0200, Herman Cremer wrote: http://www.psitek.co.za/gsm.html These guys are also in RSA, and Australia. This unit does exactly the same as the DigiCell, which mark is talking about, but is a much better product (and more expensive) maybe they export ? Except that when I've been to the USA - I've needed a 1900Mhz phone - this is only 900 and 1800... *GSM INTERFACE * GSM output 900MHz: Class 4/5, 2W EGSM * GSM output 1800MHz: Class 1, 1W DCS * SIM interface: 3V mini SIM but look at the website (Hey, it looks like my box!) as the features are what you are looking for... I believe Motorola was one of the earlier producers of this type of device - but would think that most of the manufacturers would have a similar type of unit. Push the Telephone access in disaster areas, where wire-network infrastructure is damaged point... :-) -Herman On Mon, 2005-02-28 at 17:21, Mark Elkins wrote: On Fri, 2005-02-25 at 16:29 -0700, Mr. James W. Laferriere wrote: Hello Mark , C. All , Is this device available for sale in the US ? All the digging I've only found outside US mentions of sales . Any help appreciated . JimL No idea. The Unit I have is a locally manufactured device called Digi-Cell - frmaritz (at) global.co.za is the email address on the box it came in Its probably 900Mhz GSM only - in the US - You'll need a 1900Mhz unit??? On Fri, 25 Feb 2005, Mark Elkins wrote: On Fri, 2005-02-25 at 13:46 +, C. Tomlinson wrote: Did you have to make any changes to use the premicell, or was it as simple as an outgoing landline call? I am looking into doing this as you can get deals where calls between chosen numbers are free :-) Absolutely no changes at all I did stick a Phone onto the 2-wire input of the 'PremiCell' to check that all worked - before going via Asterisk - but thats all. [part of the previous message] In South Africa, I have a 4-port ISDN BRI (Euro-ISDN) with bristuff. Calls to Cell phones are no different to any other call... I also added a Digium 4-port analogue card - and have a 'PremiCell' connected to a Trunk line. The PremiCell is a fixed cell device that gives dial-tone in the same way that a Telcom Trunk line would work - except there is no copper to he exchange - just a stubby cellphone antenna. In South Africa it is MUCH MUCH cheaper to make a Cell to Cell call than from Telcom to Cell I'm surprised that more people do not put down a 'PremiCell' type device and route all Cell calls out through it... ___ 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: http://www.psitek.co.za/gsm.html These guys are also in RSA, and Australia. This unit does exactly the same as the DigiCell, which mark is talking about, but is a much better product (and more expensive) [...] Content analysis details: (0.9 points, 5.0 required) pts rule name description -- -- 0.1 FORGED_RCVD_HELO Received: contains a forged HELO 0.8 CELL_PHONE_IMPROVE BODY: Talks about cell-phone signal improvement -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options
[Asterisk-Users] Manager Message: Originate failed being generated when callee does not pick up
I am using the Manager tooriginate calls. I am getting "Message: Originate failed" even the phone is ringing on the other end of the line. How can I reliably know if the phone on the other end of the line is receiving the call? Thanks, Tom Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do more. Manage less.___ 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] RE: Asterisk-Users Digest, Vol 7, Issue 323
I fear that list digest did not forward to me all the messages... buying cell phone adapters is quite unfeasible at this point, since the installation at hand uses 8 BRI for outgoing calls, and the customer negotiated very special rates for handling all the traffic through his voice carrier. Moreover, in italy you have 4 cell phone operators, and you should add a bunch of call phone adapters for each of them (call.operator A to cell.operator B costs much more than land operator X to any cell. operator) in another installation, with a single 4BRI card conntected to point-to-multipoint ISDN lines, everything works fine (even calls to cell phones), and land calls are fine. this is the problem that puzzles me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Monday, February 28, 2005 6:14 PM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 7, Issue 323 Date: Mon, 28 Feb 2005 18:07:42 +0200 From: Mark Elkins [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Zaptel (Junghanns 4BRI card) to cell phoneproblem To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain On Mon, 2005-02-28 at 17:50 +0200, Herman Cremer wrote: http://www.psitek.co.za/gsm.html These guys are also in RSA, and Australia. This unit does exactly the same as the DigiCell, which mark is talking about, but is a much better product (and more expensive) maybe they export ? Except that when I've been to the USA - I've needed a 1900Mhz phone - this is only 900 and 1800... *GSM INTERFACE * GSM output 900MHz: Class 4/5, 2W EGSM * GSM output 1800MHz: Class 1, 1W DCS * SIM interface: 3V mini SIM but look at the website (Hey, it looks like my box!) as the features are what you are looking for... I believe Motorola was one of the earlier producers of this type of device - but would think that most of the manufacturers would have a similar type of unit. Push the Telephone access in disaster areas, where wire-network infrastructure is damaged point... :-) -Herman On Mon, 2005-02-28 at 17:21, Mark Elkins wrote: On Fri, 2005-02-25 at 16:29 -0700, Mr. James W. Laferriere wrote: Hello Mark , C. All , Is this device available for sale in the US ? All the digging I've only found outside US mentions of sales . Any help appreciated . JimL No idea. The Unit I have is a locally manufactured device called Digi-Cell - frmaritz (at) global.co.za is the email address on the box it came in Its probably 900Mhz GSM only - in the US - You'll need a 1900Mhz unit??? On Fri, 25 Feb 2005, Mark Elkins wrote: On Fri, 2005-02-25 at 13:46 +, C. Tomlinson wrote: Did you have to make any changes to use the premicell, or was it as simple as an outgoing landline call? I am looking into doing this as you can get deals where calls between chosen numbers are free :-) Absolutely no changes at all I did stick a Phone onto the 2-wire input of the 'PremiCell' to check that all worked - before going via Asterisk - but thats all. [part of the previous message] In South Africa, I have a 4-port ISDN BRI (Euro-ISDN) with bristuff. Calls to Cell phones are no different to any other call... I also added a Digium 4-port analogue card - and have a 'PremiCell' connected to a Trunk line. The PremiCell is a fixed cell device that gives dial-tone in the same way that a Telcom Trunk line would work - except there is no copper to he exchange - just a stubby cellphone antenna. In South Africa it is MUCH MUCH cheaper to make a Cell to Cell call than from Telcom to Cell I'm surprised that more people do not put down a 'PremiCell' type device and route all Cell calls out through it... ___ 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] Manager Message: Originate failed beinggenerated when callee does not pick up
I am getting "Message: Originate failed" even the phone is ringing on the other end of the line. Originate will ring your own extension first and when you pick up, call the other number. If you don't pick up your extension, you will receive the message you see. Bill Seddon From: [EMAIL PROTECTED] on behalf of Thomas MillerSent: Mon 28/02/2005 17:33To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Manager "Message: Originate failed" beinggenerated when callee does not pick up I am using the Manager tooriginate calls. I am getting "Message: Originate failed" even the phone is ringing on the other end of the line. How can I reliably know if the phone on the other end of the line is receiving the call? Thanks, Tom Do you Yahoo!?Yahoo! Mail - 250MB free storage. Do more. Manage less. ___ 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] phpconfig
Hi, The default install from that turorial gave me fully functioning links etc. What format are your config files in; care to post an extract? What version of PHP are you running? Regards, C -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Allan hank Sent: 28 February 2005 15:23 To: Time Bandit Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] phpconfig Hello, That's the document i read and got all the relevant links. I also tried to follow all the predures . More help is appreciated, Thanks very much Allan On Mon, 28 Feb 2005 10:13:02 -0500, Time Bandit [EMAIL PROTECTED] wrote: Questions: 1) Am i using an older version? If so, where can i get a newr version? 2) Am i missing some configuration, which one? See this newly created document, it explains everything you need to make it work. http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20phpconfig It's been written with the help of peoples on this list. hth ___ 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] how to increase max number of simulatneous outgoing calls
Hello, I would like to know how to maximize the number of simultatenous outgoing calls. The application I am working on uses the Manager API to originatethe outgoing calls, and the callswill hang up one or two seconds after the callee picks up the phone. I know it sounds strange but that is what the application does itjust playbacks a one secondaudio file. There will be no "long coversations" or any nonsense like that. The Manager API is working great, but it seems to "queue" the outgoing calls and only does them one at a time through my VOIP provider (teliax.com). 1) Is the max number of simultaneous ougoing calls solely dicated by the VOIP provider, or is it limited by * ? 2) Does IAX2 have an advantage over SIP in this regard? 3) I would like to have about 50-100 simulatneous outgoing calls, what do I need to do to accomplish this? Thx, Tom Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard.___ 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] Weird behaviour on incoming DIDs
--On Monday, February 28, 2005 08:46 -0800 beonice [EMAIL PROTECTED] wrote: Folks, I have a problem here. I have 2 DIDs, one a 415 number and the other a 650 number. I have my extensions.conf set up to handle both of them exactly the same way, passing them to an internal context. When _I_ dial either DID, I get exactly the same behaviour that I have specified (the call is answered, and then I play my own welcome mesage, then handle any extension dialed). However, when one of my friends dials in, the 415 DID consistently works as designed, but the 650 DID sometimes just tells him goodbye, and then hangs up on him! I don't know if it matters, but I am calling from the 650 area code and he's calling sometimes from the 415 area code and sometimes from 408. No, there is no pattern as to which incoming call gets hung-up! Here are the relevant sections of my extensions.conf: How are these DIDs being delivered? SIP, IAX, PRI? Either way I would connect to my asterisk console in verbose mode (set verbose 255) and get someone to call and then see whats different about the failed call versus the successful calls. ___ 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] Asterisk network architecture
Hello, I'm currently working on a new installation and wondering which architecture and protocol I should use... I want to share my Asterisk server between users on my internal LAN and a user connecting via Internet... So, my server has to be reachable from outside and also from inside... For the external user, I'll use a softphone and probably IAX2 ? What about internal users and IP phone ? We have grandstream budgetone and I don't know which protocol I should use. I'd like to find a way to have my asterisk server in a DMZ protected from outside and not directly on the internal network. Is there any recommended architecture ? Also, is the traffic between external user to asterisk server encrypted ? and is there any softphone with rsa keys auth ? 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] How to charge incoming calls with ASTCC ?
I wonder if it is possible to setup exensions.conf so, that incoming calls are charged. bye Ronald ___ 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] Ring state patch
Hi Does anyone know the procedure for installing the ring state patch for snom phones . I really need this. Id appreciate any help. Geoffrey Sachs ___ 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] Asterisk network architecture
I'd like to find a way to have my asterisk server in a DMZ protected from outside and not directly on the internal network. Is there any recommended architecture ? One of my current installs is a DMZ with an * server protected from outside and inside with Monowall: http://www.m0n0.ch/wall/ The Asterisk server talks IAX over the Net to a primary Asterisk server that provides PSTN connectivity; SIP is used inside the LAN. Asterisk is sandwiched between the two monowalls and it works great. Basically, Monowall rocks. Brain dead easy install and boots off a CD; get's it's config from a user-editable XML file on a floppy. I especially like the traffic shaper which seems to work fine. My only quibble is that it is extremely bitchy about the traffic rules, they have to be set up *just* so, or they won't work. The first time I set it up, I had to mess about with a port scanner and sniffer to figure out that it was working. But once you get used to it, it's a no brainer. hth ___ 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] List tips for new subscribers
-Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] On February 23, 2005 10:21 am, [EMAIL PROTECTED] wrote: Oh I'm sorry. This is the first list I've joined where this is such a big issue! Forgive me for not having your superior understanding of mail clients, and/or list servers! You have a *servere* inferiority complex. I asked a simple question. The only people who don't see why it's a problem use inferior mail user agents which don't support threading, or perhaps they don't realize that they can do threading. FWIW, I was unaware of this issue for a long time, too. I did use threading in Outlook, but like many clients it fakes it using the subject headers. It wasn't until I started using Thunderbird elsewhere that I understood what people were complaining about. ___ 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] Dial out through Broadvoice
see bottom
- Original Message -
From: John Millican [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, February 28, 2005 10:21 AM
Subject: Re: [Asterisk-Users] Dial out through Broadvoice
On Saturday February 26 2005 4:45 pm, John Millican wrote:
On Saturday February 26 2005 4:30 pm, Chris Ford wrote:
I tried to call you number to see what I would get and you have a
verizon
Voice messaging service.
if you called the 6037862111 that is a voicemail number tyhat i was
calling
to test knowing it would not be busy and would not bother anyone.
Make sure you have your iax set up right in the Iax.conf and your
outbaound registering string going back out.
I have mine set up that I dial 6 to get out on my broadvoice line
and 9
to get out on my voice pulse line.
I am not using IAX at all. Did not think broadvoice supported it, am
I
wrong?
More Comments at BOTTOM
Hello all,
When I call the Broadvoice number all is good.
When I try to call out through DISA on my broadvoice line i get
the
following:
Executing Dial(SIP/147.135.0.129-0815bc60,
SIP/[EMAIL PROTECTED]|30) in new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 480 Temporarily Not Available back from
147.135.16.128
-- SIP/proxy.bos.broadvoice.com-3493 is circuit-busy
== Everyone is busy/congested at this time
-- Executing Busy(SIP/147.135.0.129-0815bc60, ) in new
stack
== Spawn extension (outgoing, 16037862111, 102) exited non-zero
on
'SIP/147.135.0.129-0815bc60'
Is this as simple as it seems? Broadvoice is circut busy? Can
any
one think
of any other reason I might get this message? Or do I just need
to
call
BroadVoice and complain? I have tried two different proxy's (ip's
in
/etc/hosts) and get the same error.
in extensions.conf:
[outgoing]
exten = _1NXXNXX, 1, dial(SIP/${EXTEN}
@proxy.bos.broadvoice.com,30) ;
exten = _1NXXNXX, 2, congestion() ; No answer, nothing
exten = _1NXXNXX, 102, busy() ; Busy
in sip.conf:
[general]
context=default ; Default context for incoming
calls
port=5060 ; UDP Port to bind to (SIP standard
port is 5060)
bindaddr=192.168.123.100 ; IP address to bind to
(0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound
calls
; Note: Asterisk only uses the first
host
; in SRV records
; Disabling DNS SRV lookups disables
the
; ability to place SIP calls based on
domain
; names to some other SIP users on the
Internet
register =
[EMAIL PROTECTED]:XX:[EMAIL PROTECTED]
[broadvoice1]
type=friend
username=603XXX
fromuser=603XXX
secret=XX
host=proxy.bos.broadvoice.com
fromdomain=sip.broadvoice.com
context=broadvoice
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no
nat=yes
[bv-in-1]
type=friend
host=sip.broadvoice.com
context=broadvoice
dtmfmode=inband
canreinvite=no
nat=yes
Try adding this line to sip:
insecure=very
Added insecure=very and same message
see if that helps. if not, try a standard registration string
instead
of the one broadvoice tells you to use.
Also - make sure you're using the password they sent you in an
email -
not the one you used when you signed up on their website.
Registration seems to work and shows as registered when i run sip show
registry.
I have been on the support line with broadvoice several times now and
still no
resolution, so I am askking help again, please. Below is sip show
registry
and a sip debug. Does any one have any sugestions? I followed the
instrution at http://edvina.net/broadvoice/ along with others and
sytill no
luck on outbound calls. in the sip debug it is showing the internal
ip in
the callid field. I have externip= external ip in sip .conf am i
missing
something else?
linux*CLI sip show registry
HostUsername Refresh State
sip.broadvoice.com:5060 [EMAIL PROTECTED]15 Registered
_
Did you add:
insecure=very
to your config? I don't see it posted. I'm nearly positive it needs to
be in there.
If that didn't work, did you try the simple registration string instead
of the one Broadvoice tells you to use?
number:[EMAIL PROTECTED]
I'm not exactly sure which of the above fixed the same problem for me -
but one of them did. When I did a sip show registry, it showed I was
registered, but I still couldn't make outbound calls until I did both
items above. These things may not work for you, but why not at least
try them? If they don't work, let us know you at least tried them and
it didn't work.
Are you using the password Broadvoice emailed you instead of the one you
registered on their website?
Another thing I'd try - eliminate the Broadvoice Proxy for now and try
to
RE: [Asterisk-Users] Asterisk + SER
Hey Thanks guys... But how can I use Asterisk for billing and accounting? Do you mean use the astcc module..? Please help... Thanks, Neel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Charles Wang Sent: Saturday, February 26, 2005 11:50 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk + SER Yes, I use this method too. On Sat, 26 Feb 2005 18:18:15 +0200, Yair Hakak [EMAIL PROTECTED] wrote: you do not need radius for ser and asterisk to speak to each other. if anything, i would suggest using SER for the endpoint and asterisk for the billing and accounting. -yair On Fri, 25 Feb 2005 23:32:42 -0500, Chris Ford [EMAIL PROTECTED] wrote: I just installed SER last night but if you want it ot talk to Asterisk I found that you should install FREERADIUS Server and RADIUS CLIENT. For it to function properly - Original Message - From: Nitesh Divecha [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, February 25, 2005 8:29 PM Subject: [Asterisk-Users] Asterisk + SER Hello All, Has anyone tried Asterisk with SER.? My main focus is billing and authentication of my endpoints. I want Asterisk to handle all my endpoints and SER to do billing/accounting stuff. Any help will be highly appreciated. Neel ___ 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 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 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] Suse 9.2 + CAPI Driver
On Mon, Feb 28, 2005 at 05:06:47PM -, Victor Alvarez wrote: Hello, I'm trying to install CAPI Driver for Suse 9.2 and I found the documentation for this pretty old since It refers to Suse 8.2 ( http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install ). This is especially apparent when I look at the section of these instructions for altering src.drv/makefile to replace the occurance of CARD_PATH. I tried to install the driver using the default makefile, with the final result of capi not installed - No such device or address (6) Wee, hard to read your long lines, is your line wrapper broken:) Anyway capi not installed is a message from your system that the hardware and the nedded modules are not working (together). Read the docs about capi4linux from your distri to get that up and running. You need a couple of more modules from your kernel like lsmod from my box: capidrv26612 3 b1pci 6276 1myHardWare b1dma 10728 0 [b1pci] myHardWare b1 15904 0 [b1pci b1dma] myHardWare capi 17856 8 capifs 3760 1 [capi] kernelcapi 29728 7 [capidrv b1pci capi] capiutil 14784 0 [capidrv kernelcapi] Are there any updated documents out there? Tons, I promise:) Regards, Victor. -- Tho/\/\as ___ 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] Asterisk + SER
it depends what you mean by billing and accounting. postpaid? prepaid? integrated into the dialplan or just for use later? you can use cdr_mysql or similar to dump everything into a DB and build billing apps on that, if you want as well. please read the stuff here: http://www.voip-info.org/wiki-Asterisk+billing most of the billing functions are very well documented. -yair p.s. the reason i said i would do the opposite of your suggestion is that SER is a better SIP proxy server than asterisk (it scales better, among other things). The downside is that the routing logic is more programmatic - i.e. extensions.conf is much simpler than ser.cfg, and there's also no handy nat=yes flag - you need rtp_proxy and nathelper, to get past NATs. I use asterisk as my PSTN gateway as well as handling all the dialing logic in asterisk, and SIP just takes care of registering endpoints. hope this helps. On Mon, 28 Feb 2005 10:13:20 -0800, Nitesh Divecha [EMAIL PROTECTED] wrote: Hey Thanks guys... But how can I use Asterisk for billing and accounting? Do you mean use the astcc module..? Please help... Thanks, Neel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Charles Wang Sent: Saturday, February 26, 2005 11:50 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk + SER Yes, I use this method too. On Sat, 26 Feb 2005 18:18:15 +0200, Yair Hakak [EMAIL PROTECTED] wrote: you do not need radius for ser and asterisk to speak to each other. if anything, i would suggest using SER for the endpoint and asterisk for the billing and accounting. -yair On Fri, 25 Feb 2005 23:32:42 -0500, Chris Ford [EMAIL PROTECTED] wrote: I just installed SER last night but if you want it ot talk to Asterisk I found that you should install FREERADIUS Server and RADIUS CLIENT. For it to function properly - Original Message - From: Nitesh Divecha [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, February 25, 2005 8:29 PM Subject: [Asterisk-Users] Asterisk + SER Hello All, Has anyone tried Asterisk with SER.? My main focus is billing and authentication of my endpoints. I want Asterisk to handle all my endpoints and SER to do billing/accounting stuff. Any help will be highly appreciated. Neel ___ 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 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 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] FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format
Kristian Kielhofner schrieb: Bastian Schern wrote: Hello, I've got problems to install zaptel on a SuSE 9.1 System. The System has got a Linux 2.6.9 Kernel. If I try to load zaptel framework (modprobe zaptel) I get this message: FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format How I can fix this. At compile time, there were no Errors. Regards Bastian Did you do a make linux26 in the zaptel directory? Yes, of course. Bastian ___ 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] Manager Message: Originate failed beinggenerated when callee does not pick up
Thanks for your help. From what you said it looks like
I should not use Originate, but there is no
alternative to the Originate action if I just want
to make an outgoing call is there?
This is what my code is sending to the Manager API:
clientSocket.Send(Encoding.ASCII.GetBytes(Action:
Originate\r\nChannel: + asteriskVoipChannel + / +
PhoneNumber + \r\nContext: justring\r\nCallerID: +
callerId + \r\nExtension: justring\r\nPriority:
1\r\n\r\nAction: Logoff\r\nActionId: 1\r\n\r\n));
Here is the extension:
[justring]
exten =
_1XX,1,DIAL(IAX2/[EMAIL PROTECTED]/${EXTEN},10,S(0))
exten = _1XX,2,Hangup
exten = s,1,Hangup
How do I use the Manager API to place an outgoing call
and reliably get a status code indicating the the call
was placed successfully?
--- Bill Seddon [EMAIL PROTECTED] wrote:
I am getting Message: Originate failed even the
phone is ringing on the other end of the line.
Originate will ring your own extension first and
when you pick up, call the other number. If you
don't pick up your extension, you will receive the
message you see.
Bill Seddon
__
Do you Yahoo!?
Yahoo! Mail - Helps protect you from nasty viruses.
http://promotions.yahoo.com/new_mail
___
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] List tips for new subscribers
David Brodbeck wrote: -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] On February 23, 2005 10:21 am, [EMAIL PROTECTED] wrote: Oh I'm sorry. This is the first list I've joined where this is such a big issue! Forgive me for not having your superior understanding of mail clients, and/or list servers! You have a *servere* inferiority complex. I asked a simple question. The only people who don't see why it's a problem use inferior mail user agents which don't support threading, or perhaps they don't realize that they can do threading. FWIW, I was unaware of this issue for a long time, too. I did use threading in Outlook, but like many clients it fakes it using the subject headers. It wasn't until I started using Thunderbird elsewhere that I understood what people were complaining about. I think the Digium listserv should just reject HTML messages and messages with multiple mailing list footers. Would cut down on a lot of traffic. 8-) --Eric ___ 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] phpconfig
The key to getting the menu entries to appear on the pages is the fgetc/fgets edits. It caught me out until I read through the code. BTW - how did that error get into CVS anyway!! Take another look at the tutorial again to see if you have missed anything else. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 266.5.1 - Release Date: 27/02/2005 ___ 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] Dial out through Broadvoice
On Monday February 28 2005 1:17 pm, Roger Hanson wrote:
see bottom
snip
Hello all,
When I call the Broadvoice number all is good.
When I try to call out through DISA on my broadvoice line i get
the
following:
Executing Dial(SIP/147.135.0.129-0815bc60,
SIP/[EMAIL PROTECTED]|30) in new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 480 Temporarily Not Available back from
147.135.16.128
-- SIP/proxy.bos.broadvoice.com-3493 is circuit-busy
== Everyone is busy/congested at this time
-- Executing Busy(SIP/147.135.0.129-0815bc60, ) in new
stack
== Spawn extension (outgoing, 16037862111, 102) exited non-zero
on
'SIP/147.135.0.129-0815bc60'
snip
in extensions.conf:
[outgoing]
exten = _1NXXNXX, 1, dial(SIP/${EXTEN}
@proxy.bos.broadvoice.com,30) ;
exten = _1NXXNXX, 2, congestion() ; No answer, nothing
exten = _1NXXNXX, 102, busy() ; Busy
in sip.conf:
[general]
context=default ; Default context for incoming calls
port=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=192.168.123.100 ; IP address to bind to (0.0.0.0 binds to
all)
srvlookup=yes ;
register
=[EMAIL PROTECTED]:XX:[EMAIL PROTECTED]
[broadvoice1]
type=friend
username=603XXX
fromuser=603XXX
secret=XX
host=proxy.bos.broadvoice.com
fromdomain=sip.broadvoice.com
context=broadvoice
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no
nat=yes
Try adding this line to sip:
insecure=very
Added insecure=very and same message
see if that helps. if not, try a standard registration string
instead
of the one broadvoice tells you to use.
Also - make sure you're using the password they sent you in an
email -
not the one you used when you signed up on their website.
Registration seems to work and shows as registered when i run sip show
registry.
I have been on the support line with broadvoice several times now and
still no
resolution, so I am askking help again, please. Below is sip show
registry
and a sip debug. Does any one have any sugestions? I followed the
instrution at http://edvina.net/broadvoice/ along with others and
sytill no
luck on outbound calls. in the sip debug it is showing the internal
ip in
the callid field. I have externip= external ip in sip .conf am i
missing
something else?
linux*CLI sip show registry
HostUsername Refresh State
sip.broadvoice.com:5060 [EMAIL PROTECTED]15 Registered
_
Did you add:
insecure=very
to your config? I don't see it posted. I'm nearly positive it needs to
be in there.
If that didn't work, did you try the simple registration string instead
of the one Broadvoice tells you to use?
number:[EMAIL PROTECTED]
I'm not exactly sure which of the above fixed the same problem for me -
but one of them did. When I did a sip show registry, it showed I was
registered, but I still couldn't make outbound calls until I did both
items above. These things may not work for you, but why not at least
try them? If they don't work, let us know you at least tried them and
it didn't work.
Are you using the password Broadvoice emailed you instead of the one you
registered on their website?
Another thing I'd try - eliminate the Broadvoice Proxy for now and try
to get working so you can make an outbound call. Then try messing with
the proxy.
_
A previous post I made (found searching the mailing list) shows:
Here's my portion of sip.conf
[952nnn]
type=friend
secret=passwordhere
regexten=952nnn
insecure=very
host=sip.broadvoice.com
fromuser=952nnn
fromdomain=sip.broadvoice.com
dtmfmode=inband
context=from-pstn
canreinvite=yes
Remember, the password you need to use is not the one you signed up with
on the broadvoice website - it's the one they emailed you.
I use a standard register string:
register=952nnn:[EMAIL PROTECTED]
I can tx/rx calls over Broadvoice just fine.
asterisk*CLI sip show registry
HostUsername Refresh State
sip.broadvoice.com:5060 952nnn 15 Registered
asterisk*CLI
I added insecure=very and went with the standard registration(as mentioned
above) and still the same error.
I added sip.broadvoice.com to the /etc/hosts, same error.
pointed sip.broadvoice.com in hosts to proxy.mia.broadvoice.com's ip, same
error. tried all the other listed proxy's, no dial out.
I am totaly stumped. Am i not providing some helpfull info? If not tell me
what i am missing and i will get it. I am sure I have
[Asterisk-Users] Advanced Conferencing options with out-of-tree modules?
I've been poking at setting up a proof-of-concept * server as a replacement for our commercial conferencing solution. I've been through the wiki and list archives, and think I have found a combination that provides the features we want/need. The combination of applications CBMysql and MeetMe2 seem to address our goals. I have MeetMe2 working. CBMysql is another story, the code looks simple enough and has been modified to leverage MeetMe2, but * restarts everytime it tries to launch CBMysql. I cannot find any examples of how to launch it from the dial plan, nor have I been able to get any meaningful debug logs. Has anyone used the CBMysql application and can provide pointers on the how to launch it? * is version 1.0.5 Many thanks, Dan ___ 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] Recommendation for dialplan in case of DDoS atta cks?
I'm trying to formulate a strategy for our interconnected Asterisk IAX peers to failover to the PSTN in the event of a DDoS. We currently use them like this: DID---PRI---Primary Asterisk---IAX---On-site Asterisk---SIP This works fine, and everyone is happy. One of my concerns, though, is if we get DDoS'd - which happens probably once every couple of years. I'd like to have the dialplan failover to PSTN to shunt calls to the PSTN---User's cell number in the case of a DDoS attack. My current thinking is K.I.S.S - just put the user's cell as the next step in the dialplan. However, I'd like for this to be controllable - when things are working OK, I don't want the calls being routed to the cells *at all*. I also don't want to have an extensions.conf and an extensions_emergency.conf and do the _emergency as an commented out include. I'd like for it to be automatic i.e. Asterisk detects Internet latency is above a certain threshold, then automagically does the cell thing. Any suggestions? I fooled around in Google for about a half hour on this, and of course the Wiki was no help. TIA ___ 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] Recommendation for dialplan in case of DDoS atta cks?
Colin Anderson wrote: I'm trying to formulate a strategy for our interconnected Asterisk IAX peers to failover to the PSTN in the event of a DDoS. We currently use them like this: DID---PRI---Primary Asterisk---IAX---On-site Asterisk---SIP This works fine, and everyone is happy. One of my concerns, though, is if we get DDoS'd - which happens probably once every couple of years. I'd like to have the dialplan failover to PSTN to shunt calls to the PSTN---User's cell number in the case of a DDoS attack. My current thinking is K.I.S.S - just put the user's cell as the next step in the dialplan. However, I'd like for this to be controllable - when things are working OK, I don't want the calls being routed to the cells *at all*. I also don't want to have an extensions.conf and an extensions_emergency.conf and do the _emergency as an commented out include. I'd like for it to be automatic i.e. Asterisk detects Internet latency is above a certain threshold, then automagically does the cell thing. Any suggestions? I fooled around in Google for about a half hour on this, and of course the Wiki was no help. TIA How about a combination of GotoIF, and app_dbodbc (or app_db): exten = 700,1,playback(ddos-on) exten = 700,2,DBput(DDOS/yes) exten = 701,1,playback(ddos-off) exten = 701,2,DBdel(DDOS/yes) [mymainaa] exten = s,1,DBGET(TRUE=DDOS/yes) exten = s,2,Do this exten =) s,102,do something else Just a very lazy, simple example, but it should work. -- Kristian Kielhofner ___ 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] Recommendation for dialplan in case of DDoS atta cks?
Primary * box detects DD0S - runs: asterisk -rx database put PANIC DDOS YES and have your dialplan look for that database family/key being set to determine which path it takes. When the primary * box detects that the DD0S is over - runs: asterisk -rx database del PANIC DDOS On Tue, 2005-03-01 at 06:40, Colin Anderson wrote: I'm trying to formulate a strategy for our interconnected Asterisk IAX peers to failover to the PSTN in the event of a DDoS. We currently use them like this: DID---PRI---Primary Asterisk---IAX---On-site Asterisk---SIP This works fine, and everyone is happy. One of my concerns, though, is if we get DDoS'd - which happens probably once every couple of years. I'd like to have the dialplan failover to PSTN to shunt calls to the PSTN---User's cell number in the case of a DDoS attack. My current thinking is K.I.S.S - just put the user's cell as the next step in the dialplan. However, I'd like for this to be controllable - when things are working OK, I don't want the calls being routed to the cells *at all*. I also don't want to have an extensions.conf and an extensions_emergency.conf and do the _emergency as an commented out include. I'd like for it to be automatic i.e. Asterisk detects Internet latency is above a certain threshold, then automagically does the cell thing. Any suggestions? I fooled around in Google for about a half hour on this, and of course the Wiki was no help. TIA ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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
