Re: [asterisk-users] DTMF input while waiting in queue...
No problem Louis...Even though in recent times I've been kind of a jerk about people not reading the documentation, I've been trying to return to my original personality on this list, a helpful member of the community. :-[ On 3/29/2011 12:47 AM, Louis Carreiro wrote: Wow... completely missed that. It was right there in the text. Sorry and thanks Sherwood! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sherwood McGowan Sent: Monday, March 28, 2011 11:07 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DTMF input while waiting in queue... On 3/28/2011 7:54 AM, Louis Carreiro wrote: Hey all! I'm trying to figure out how to have a queue accept an inbound caller's key press to action on. At first I'm just trying to implement a Press 1 to leave a voice mail announced and at any time in the queue, the user can press 1 and go to the queue's voicemail. Later I'd like to have it accept Press 1 if this is an x issue, press 2 if this a y problem and I'll have UserEvent's generated for the press. *snip* In your queues.conf, in the definition for 1820, add the following: context=queue1820-exit Then, in your dialplan create a new context: [queue1820-exit] exten = 1,1,Noop(Caller Pressed 1 to leave a voicemail) exten = 1,n,Voicemail(voicemailbox,theoptionsyouwant) exten = 1,n,Hangup That should get you started...Read about the context configuration option here: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf Cheers! -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan help: hang up incoming call and call the number back
design your dial-plan for routing a specific number to different context , you can try func_odbc for query to DB if you have a large number of setup. ideally its called click to call but you are made it as, miss call and you will get a call. regards dhaval On Mon, Mar 28, 2011 at 5:21 PM, Roger Burton West ro...@firedrake.orgwrote: On Mon, Mar 28, 2011 at 05:14:50PM +0530, Raj Mathur wrote: Is there a better way of handling the post-hangup processing? Callfiles? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] disconnecting destination channel
Dear All I am using Asterisk 1.4.17 in a calling card application. Following description explains the usage: A call/request hits asterisk from an ip xxx.xxx.xxx.xxx which opens a channel for this ip (Lets call it Channel A). Asterisk answers the call and play IVRs first asking the PIN and then destination number in an AGI making use of radius server for authentication/authorization. Once done with that, Asterisk uses 'Dial' application to move forward sending it to ip yyy.yyy.yyy.yyy which opens another channel to this ip (lets call it channel B). I want to include '##' feature (for making another call), when caller presses ## during a call, destination channel B is hanged up (i need to hangup the destination channel to billing purposes) keeping originating channel A alive and another AGI application is triggered for caller to enter another destination number. Im using feature.conf's application map something like below: followupcall = ##,peer/caller,Hangup At least feature is working but hanging up both channels...Problem!!! If there is any other application to hang up the destination channel, what is that? Also what is the status of originating channel? Where should the call to second AGI be put in the dial plan? I hope you guys understand my problem/issue. Please guide me, thanks alot in advance. -- Best Regards Atif Razzaq http://atif-razzaq.blogspot.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Checking status of a cell phone
On Tue, 29 Mar 2011 07:48:08 +0200, magnu...@inputinterior.se wrote: I was a little unclear, it is not the cell phone that does the call-back, it is the cell-phone-network. Makes more sense :-) Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Transfer Extensions
Hi All, I am having some issues with Asterisk 1.8.3 extensions with a SIP Phone and an gateway. My setup is that I have my SIP Phone setup to register with the gateway. Then the gateway should sent calls to the Asterisk as a type of friend. This works fine if the SIP Phone configuration username and password isn't already set into the asterisk. The configuration of the SIP Phone username and password on the asterisk is just for backup/testing purpose. The situation now is that when the SIP Phone is registered to the gateway and I make a call from the SIP Phone to the gateway, which sends the calls to the Asterisk. If the username in the From: of the SIP messages matches any peers in the realtime configuration of the sipfriends. The asterisk will automatically use the From field and take that context, which in this case is the context of testing. While the actual call came from the gateway which has a context of outgoing-calls. Is there a way to of not transfer or to limit the context to the peer that the call actually came from? Gateway = context(outgoing-call) SIP Phone = context(testing) Asterisk using realtime config with a sipfriend of type friend username=123 Call make from SIP Phone (username=123) to gateway, which sends to asterisk. The asterisk would use the context testing instead of from outgoing-call, since I assume that it is using the From field in the SIP Message. Please help. Many thanks. Regards, Kengie -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan help: hang up incoming call and call the number back
On Tue, Mar 29, 2011 at 12:23 PM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: design your dial-plan for routing a specific number to different context , you can try func_odbc for query to DB if you have a large number of setup. ideally its called click to call but you are made it as, miss call and you will get a call. On Mon, Mar 28, 2011 at 5:21 PM, Roger Burton West ro...@firedrake.org wrote: On Mon, Mar 28, 2011 at 05:14:50PM +0530, Raj Mathur wrote: Is there a better way of handling the post-hangup processing? Callfiles? Thanks, call files worked beautifully. Takes a couple of commands to make the call file in Asterisk (I didn't want to call any heavy external programs like Perl or Awk, though that would have been more elegant), and the rest works out of the box. Regards, -- Raj -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 Packages for Debian and Ubuntu
- upgrade policy - is it intended that someone who has Debian 6 with the existing Asterisk 1.6 packages (from Debian's maintainer) can just upgrade to the Digium package without moving or changing any config? There is nothing specific about the packages that is going to make this situation any better or worse than any method of upgrading from Asterisk 1.6.X to Asterisk 1.8. Issues related to version compatibility can be found in the UPGRADE*.txt files in the Asterisk source. http://svn.asterisk.org/view/asterisk/trunk/UPGRADE-1.8.txt?view=markup Apart from the 1.8 release notes though, there is no need to do any specific changes when going from the Debian-maintained 1.6 package to the Digium-maintained 1.8? I tried the packages (clean install) on one machine yesterday and I noticed that they depend on some of the asterisk packages within the Debian archive, while other packages get pulled down from the Digium archive. Is that intended? I tried to do another machine today and found that your key has gone missing from the key server: # apt-key adv --keyserver subkeys.pgp.net --recv-keys 175E41DF Executing: gpg --ignore-time-conflict --no-options --no-default-keyring --secret-keyring /etc/apt/secring.gpg --trustdb-name /etc/apt/trustdb.gpg --keyring /etc/apt/trusted.gpg --primary-keyring /etc/apt/trusted.gpg --keyserver subkeys.pgp.net --recv-keys 175E41DF gpg: requesting key 175E41DF from hkp server subkeys.pgp.net gpgkeys: key 175E41DF not found on keyserver gpg: no valid OpenPGP data found. gpg: Total number processed: 0 It was definitely there when I tried it yesterday - has it been revoked or something? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
On Mon, 28 Mar 2011 08:20:23 -0400, vip killa vipki...@gmail.com wrote: Is anyone using asterisk with fail2ban? Sorry for hi-jacking the thread, but I was wondering if there were a lighter alternative that I could run on appliances? Python uses too much RAM, but I need to find a way to ban hackers from trying to connect to Asterisk from the Net. Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
On Mon, 28 Mar 2011 08:20:23 -0400, vip killa vipki...@gmail.com wrote: Is anyone using asterisk with fail2ban? Sorry for hi-jacking the thread, but I was wondering if there were a lighter alternative that I could run on appliances? Python uses too much RAM, but I need to find a way to ban hackers from trying to connect to Asterisk from the Net. I had worked with the sshguard guys to add support for Asterisk; I believe they added basic support. I haven't gotten around to revisiting that issue just yet so I don't know for sure. http://lists.digium.com/pipermail/asterisk-users/2010-December/256928.html sshguard is *extremely* lightweight compared to most things; it's a very efficient compiled C application that doesn't have (m?)any dependencies. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 Packages for Debian and Ubuntu
On Tue, Mar 29, 2011 at 01:59:54PM +0200, Daniel Pocock wrote: - upgrade policy - is it intended that someone who has Debian 6 with the existing Asterisk 1.6 packages (from Debian's maintainer) can just upgrade to the Digium package without moving or changing any config? There is nothing specific about the packages that is going to make this situation any better or worse than any method of upgrading from Asterisk 1.6.X to Asterisk 1.8. Issues related to version compatibility can be found in the UPGRADE*.txt files in the Asterisk source. http://svn.asterisk.org/view/asterisk/trunk/UPGRADE-1.8.txt?view=markup Apart from the 1.8 release notes though, there is no need to do any specific changes when going from the Debian-maintained 1.6 package to the Digium-maintained 1.8? I tried the packages (clean install) on one machine yesterday and I noticed that they depend on some of the asterisk packages within the Debian archive, while other packages get pulled down from the Digium archive. Is that intended? I tried to do another machine today and found that your key has gone missing from the key server: # apt-key adv --keyserver subkeys.pgp.net --recv-keys 175E41DF Executing: gpg --ignore-time-conflict --no-options --no-default-keyring --secret-keyring /etc/apt/secring.gpg --trustdb-name /etc/apt/trustdb.gpg --keyring /etc/apt/trusted.gpg --primary-keyring /etc/apt/trusted.gpg --keyserver subkeys.pgp.net --recv-keys 175E41DF gpg: requesting key 175E41DF from hkp server subkeys.pgp.net gpgkeys: key 175E41DF not found on keyserver gpg: no valid OpenPGP data found. gpg: Total number processed: 0 The key does not have to be there for you to trust it (anybody can upload a key there). Though having it there (and having it signed by more people) would naturally help. Another issue: the package name is different 'asterisk-1.8' vs. 'asterisk'. It has: Package: asterisk-1.8 Provides: asterisk Conflicts: asterisk (1.8.0) Whereas other packages are named 'asterisk-*' (asterisk-h323, asterisk-dahdi, etc.). So it deviates from the Debian naming convention. And would still not allow co-install with a future asterisk-1.10 package. Also note http://www.debian.org/doc/debian-policy/ch-relationships.html#s-replaces -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Get phone number from SIP header PAI
Hello list,
I want to get the phone number out of the following P-Asserted-Identity
header :
/BlaBlaBla sip://88779922//@192.168.8.10;user=phone/
I do the following in the dialplan :
/exten = _XXX.,n,Set(PY=${SIP_HEADER(P-Asserted-Identity)})
exten = _XXX.,n,Set(PY2=${CUT(PY,@,1)})/
This gives me :
/BlaBlaBla sip://88779922/
How can I extract /88779922/ out of this string ??
I'm trying this :
/exten = _XXX.,n,Set(PY4=${CUT(PY2,\:,1-)}) /
but this does not change a thing to the string...
I just want everything after the comma...
Kind regards,
Jonas.
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Re: [asterisk-users] asterisk and fail2ban
On Tue, 29 Mar 2011 07:31:18 -0500 (CDT), Joe Greco jgr...@ns.sol.net wrote: sshguard is *extremely* lightweight compared to most things; it's a very efficient compiled C application that doesn't have (m?)any dependencies. Thanks much for the tip. I'll study how to install/configure iptable and sshguard. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Debugging not going to log file
I have an Asterisk server running 1.6.2.13, where I can't seem to get the increased logging to save to the /var/log/asterisk/messages file. I have tried using the standard core set debug 10 and core set verbose 10, as well as specifically pointing it to the filename with core set debug 10 /var/log/asterisk/messages. Still, only the most serious errors are being reported to the messages log file. It seems to work fine with my other Asterisk running 1.4.23.1. Is there something else that I'm missing? Dean Hoover Waukesha, Wisconsin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Debugging not going to log file
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dean Hoover Sent: Tuesday, March 29, 2011 9:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Debugging not going to log file I have an Asterisk server running 1.6.2.13, where I can't seem to get the increased logging to save to the /var/log/asterisk/messages file. I have tried using the standard core set debug 10 and core set verbose 10, as well as specifically pointing it to the filename with core set debug 10 /var/log/asterisk/messages. Still, only the most serious errors are being reported to the messages log file. It seems to work fine with my other Asterisk running 1.4.23.1. Is there something else that I'm missing? Dean Hoover Waukesha, Wisconsin Check your logger.conf - the messages line probably says error instead of notice, warning, error, debug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Debugging not going to log file
On Tue, Mar 29, 2011 at 11:05 AM, Dean Hoover kb7...@gmail.com wrote: I have an Asterisk server running 1.6.2.13, where I can't seem to get the increased logging to save to the /var/log/asterisk/messages file. I have tried using the standard core set debug 10 and core set verbose 10, as well as specifically pointing it to the filename with core set debug 10 /var/log/asterisk/messages. Still, only the most serious errors are being reported to the messages log file. It seems to work fine with my other Asterisk running 1.4.23.1. Is there something else that I'm missing? Dean Hoover Waukesha, Wisconsin I had this happen a month ago, don't feel bad... In http://svn.asterisk.org/svn/asterisk/branches/1.4/configs/logger.conf.sample check for debug on the end of the logging method. ;debug = debug console = notice,warning,error ;console = notice,warning,error,debug --- Look here messages = notice,warning,error ;full = notice,warning,error,debug,verbose -- ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Debugging not going to log file
That did it. Thanks everyone! On Tue, Mar 29, 2011 at 9:11 AM, Andrew Latham lath...@gmail.com wrote: On Tue, Mar 29, 2011 at 11:05 AM, Dean Hoover kb7...@gmail.com wrote: I have an Asterisk server running 1.6.2.13, where I can't seem to get the increased logging to save to the /var/log/asterisk/messages file. I have tried using the standard core set debug 10 and core set verbose 10, as well as specifically pointing it to the filename with core set debug 10 /var/log/asterisk/messages. Still, only the most serious errors are being reported to the messages log file. It seems to work fine with my other Asterisk running 1.4.23.1. Is there something else that I'm missing? Dean Hoover Waukesha, Wisconsin I had this happen a month ago, don't feel bad... In http://svn.asterisk.org/svn/asterisk/branches/1.4/configs/logger.conf.sample check for debug on the end of the logging method. ;debug = debug console = notice,warning,error ;console = notice,warning,error,debug --- Look here messages = notice,warning,error ;full = notice,warning,error,debug,verbose -- ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Removing Polycom Transfer Softkey
Sorry, for some reason I misread it as the forward feature. On Sun, Mar 27, 2011 at 9:16 PM, Mark Murawski markm-li...@intellasoft.net wrote: From the polycom pdf: divert.fwd.x.enabled If set to 1, the user will be able to enable universal call forwarding through the soft key menu. This sounds like it turns on and turns off the call forwarding feature on the phone. I can try it out Monday, but I don't see where it has any relation to transfer (both attended and blind). On 03/27/2011 08:43 PM, C F wrote: In phone.cfg set the following line to divert.fwd.1.enabled=0 from: divert.fwd.1.enabled=1 For more info check page 323: http://supportdocs.polycom.com/PolycomService/support/global/documents/support/setup_maintenance/products/voice/spip_ssip_vvx_Admin_Guide_SIP_3_2_2_eng.pdf On Fri, Mar 25, 2011 at 6:33 PM, Mark Murawski markm-li...@intellasoft.net wrote: Sorry for the crosspost. This was supposed to be on -users I know some of you are polycom gurus... Anyone know how to remove transfer from a polycom 33x phone? We've set allowtransfer=no, but we would like to remove a polycom soft key as well. -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Removing Polycom Transfer Softkey
Look at page 311 in that manual If you disable the soft keys and then reassign the hard key it should - at least in theory - be possible to accomplish. On Tue, Mar 29, 2011 at 11:46 AM, C F shma...@gmail.com wrote: Sorry, for some reason I misread it as the forward feature. On Sun, Mar 27, 2011 at 9:16 PM, Mark Murawski markm-li...@intellasoft.net wrote: From the polycom pdf: divert.fwd.x.enabled If set to 1, the user will be able to enable universal call forwarding through the soft key menu. This sounds like it turns on and turns off the call forwarding feature on the phone. I can try it out Monday, but I don't see where it has any relation to transfer (both attended and blind). On 03/27/2011 08:43 PM, C F wrote: In phone.cfg set the following line to divert.fwd.1.enabled=0 from: divert.fwd.1.enabled=1 For more info check page 323: http://supportdocs.polycom.com/PolycomService/support/global/documents/support/setup_maintenance/products/voice/spip_ssip_vvx_Admin_Guide_SIP_3_2_2_eng.pdf On Fri, Mar 25, 2011 at 6:33 PM, Mark Murawski markm-li...@intellasoft.net wrote: Sorry for the crosspost. This was supposed to be on -users I know some of you are polycom gurus... Anyone know how to remove transfer from a polycom 33x phone? We've set allowtransfer=no, but we would like to remove a polycom soft key as well. -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spa8000 spa2102 t38 faxing
t38 from the itsp to asterisk using FFA = works t38 from the spa8000 to the local asterisk using FFA = works (no differance if FAX Passthru Method is set to reinvite or nse) g711 to a remote system = works(i just turned off t38 on the spa8000) just the t38 pt is not working the itsp is using sonus switches t38pt_udptl=yes,redundancy,maxdatagram=400 is set on the itsp i couldnt yet get a normal trace as the system was very busy and lots of debug info i'll try todo it tonight when the system is almost quiet btw is there any variable to check if the channel is using t38 ? Thanks On Mon, Mar 28, 2011 at 1:10 AM, Larry Moore lmo...@starwon.com.au wrote: On 28/03/2011 5:48 AM, Israel Gottlieb wrote: still no luck i hear it change to t38 but it just doesnt connect Do you have two fax devices at your end, even a fax-modem attached to a computer will do? You are going to need to provide more information such as your current configuration and traces of the sessions. If you turn off all T.38 options in Asterisk and on the SPA you should still be able to make a transmissions using the G711 codecs. Can you confirm you are able to send a facsimile from your device using a PSTN line? Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get phone number from SIP header PAI
On Tue, Mar 29, 2011 at 8:19 AM, Jonas Kellens jonas.kell...@telenet.bewrote:
Hello list,
snip
I'm trying this :
*exten = _XXX.,n,Set(PY4=${CUT(PY2,\:,1-)}) *
but this does not change a thing to the string...
Try the following:
*exten = _XXX.,n,Set(PY4=${CUT(PY2,:,2)}) *
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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Re: [asterisk-users] asterisk and fail2ban
On 3/29/2011 7:16 AM, Gilles wrote: On Mon, 28 Mar 2011 08:20:23 -0400, vip killa vipki...@gmail.com wrote: Is anyone using asterisk with fail2ban? Sorry for hi-jacking the thread, but I was wondering if there were a lighter alternative that I could run on appliances? Python uses too much RAM, but I need to find a way to ban hackers from trying to connect to Asterisk from the Net. Thank you. First thing I'd do is restrict the ip blocks your sip endpoints can register/call from in sip.conf (or your database's table for sip endpoints) -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: First thing I'd do is restrict the ip blocks your sip endpoints can register/call from in sip.conf (or your database's table for sip endpoints) Thanks for the idea, but it's not possible, as the Asterisk must be accessible for road warriors and receive SIP calls from anyone. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan First thing I'd do is restrict the ip blocks your sip endpoints can register/call from in sip.conf (or your database's table for sip endpoints) On Tue, 29 Mar 2011, Gilles wrote: Thanks for the idea, but it's not possible, as the Asterisk must be accessible for road warriors and receive SIP calls from anyone. Really? How many callers are you expecting from North Korea, Libya, China, Iran, etc? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR MYSQL missing field data
Thanks again Tilghman OK now I am back to the original. [columns] ;static value = column commented now ;aliascdrvar = columncommented now These are bogus and should never have been uncommented. alias start = calldate uncommented now alias callerid = clid uncommented now Rest are all back to commented ;alias src = src ;alias dst = dst ;alias dcontext = dcontext ;alias channel = channel ;alias dstchannel = dstchannel ;alias lastapp = lastapp ;alias lastdata = lastdata ;alias duration = duration ;alias billsec = billsec ;alias disposition = disposition ;alias amaflags = amaflags ;alias accountcode = accountcode ;alias userfield = userfield ;alias uniqueid = uniqueid I restarted Asterisk and I still have all the fields outgoing and incoming except for the CLID field. Clid is populated in the CSV Simple file as well as the CSV Custom file. -- Tilghman -- Eric W. Davenport Cert-In Software Systems, Inc. P.O. Box 346 Bakersville, NC 28705 800-873-0110 ewdavenp...@certin.com www.certin.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
On 3/29/2011 12:25 PM, Steve Edwards wrote: On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan First thing I'd do is restrict the ip blocks your sip endpoints can register/call from in sip.conf (or your database's table for sip endpoints) On Tue, 29 Mar 2011, Gilles wrote: Thanks for the idea, but it's not possible, as the Asterisk must be accessible for road warriors and receive SIP calls from anyone. Really? How many callers are you expecting from North Korea, Libya, China, Iran, etc? Thanks Steve, you just emailed exactly what I was going to say... Remember guys, there's a LOT of IP blocks out there that are almost definitely not going to be somewhere you expect to receive SIP traffic from. Where are you located? Where do your road warriors usually travel? Start by blocking countries that are not going to be expected to send traffic 98% of the time. When I first started out as a consultant, I helped get a certain U.S. ITSP up and running, and we reduced fraud and hack attempts DRASTICALLY simply by blocking most of the countries that are pretty much known for the prolific numbers of hackers. Sure, we had like, 2 customers call in to say they had traveled abroad (or sent their device to a family/friend abroad) and couldn't get their device to register. But seriously, it was rare. Either way, just a suggestion -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
On Tue, 29 Mar 2011 12:34:04 -0500, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: Remember guys, there's a LOT of IP blocks out there that are almost definitely not going to be somewhere you expect to receive SIP traffic from. I agree. Is there a list I could use to check which blocks have been allocated to which countries so I can add them to Asterisk's blacklist? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
On Tue, Mar 29, 2011 at 2:34 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On 3/29/2011 12:25 PM, Steve Edwards wrote: On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan First thing I'd do is restrict the ip blocks your sip endpoints can register/call from in sip.conf (or your database's table for sip endpoints) On Tue, 29 Mar 2011, Gilles wrote: Thanks for the idea, but it's not possible, as the Asterisk must be accessible for road warriors and receive SIP calls from anyone. Really? How many callers are you expecting from North Korea, Libya, China, Iran, etc? Thanks Steve, you just emailed exactly what I was going to say... Remember guys, there's a LOT of IP blocks out there that are almost definitely not going to be somewhere you expect to receive SIP traffic from. Where are you located? Where do your road warriors usually travel? Start by blocking countries that are not going to be expected to send traffic 98% of the time. When I first started out as a consultant, I helped get a certain U.S. ITSP up and running, and we reduced fraud and hack attempts DRASTICALLY simply by blocking most of the countries that are pretty much known for the prolific numbers of hackers. Sure, we had like, 2 customers call in to say they had traveled abroad (or sent their device to a family/friend abroad) and couldn't get their device to register. But seriously, it was rare. Either way, just a suggestion -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant First step should be on the AS level. If you do not have access to advertised networks then use http://www.spamhaus.org/drop/ The Spamhaus Don't Route Or Peer List and the script in the FAQ. -- ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
On 3/29/2011 12:42 PM, Gilles wrote: On Tue, 29 Mar 2011 12:34:04 -0500, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: Remember guys, there's a LOT of IP blocks out there that are almost definitely not going to be somewhere you expect to receive SIP traffic from. I agree. Is there a list I could use to check which blocks have been allocated to which countries so I can add them to Asterisk's blacklist? http://www.maxmind.com/app/ip-location -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] wrong from URI in options message
I recently configured a SIP peer which i must specify my fromuser as my ten digit DID. I send calls to this peer, but whenever Asterisk sends an options message, the fromuser is asterisk. Is this a bug? Or is there some other config I must make ? register = 211941:123456@10.0.138.226/211941~600 [peer](!) type=peer context=inbound qualify=yes qualifyfreq=300 insecure=port,invite nat=yes outgoinglimit=4 incominglimit=4 [mypeer](peer) host=10.0.138.226 defaultuser=211941 fromuser=211941 md5secret=023f30a320a5781e8ffd1af9888012af incominglimit=10 IP (tos 0x0, ttl 64, id 9242, offset 0, flags [none], proto UDP (17), length 555) 10.0.1.3.5060 10.0.138.226.5060: SIP, length: 527 OPTIONS sip:10.0.138.226 SIP/2.0 Via: SIP/2.0/UDP 10.0.83.61:5060;branch=z9hG4bK6abb74e3;rport Max-Forwards: 70 From: asterisk sip:asterisk@10.0.83.61;tag=as7444eb08 To: sip:10.0.138.226 Contact: sip:asterisk@10.0.83.61:5060 Call-ID: 20afd7e40fb31362355eae245dae1fd6@10.0.83.61:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.2.3 Date: Tue, 29 Mar 2011 17:43:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 IP (tos 0xb8, ttl 250, id 0, offset 0, flags [none], proto UDP (17), length 411) 10.0.138.226.5060 10.0.1.3.5060: SIP, length: 383 SIP/2.0 403 From: URI not recognized Via: SIP/2.0/UDP 10.245.83.61:5060;received=10.0.83.61;branch=z9hG4bK6abb74e3;rport=5060 From: asterisk sip:asterisk@10.0.83.61;tag=as7444eb08 To: sip:10.0.138.226;tag=metaswitch+1+0+e288612a Call-ID: 20afd7e40fb31362355eae245dae1fd6@10.0.83.61:5060 CSeq: 102 OPTIONS Server: DC-SIP/2.0 Organization: Content-Length: 0 -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wrong from URI in options message
On 3/29/2011 12:52 PM, Jeremy Kister wrote: I recently configured a SIP peer which i must specify my fromuser as my ten digit DID. I send calls to this peer, but whenever Asterisk sends an options message, the fromuser is asterisk. Is this a bug? Or is there some other config I must make ? register = 211941:123456@10.0.138.226/211941~600 [peer](!) type=peer context=inbound qualify=yes qualifyfreq=300 insecure=port,invite nat=yes outgoinglimit=4 incominglimit=4 [mypeer](peer) host=10.0.138.226 defaultuser=211941 fromuser=211941 md5secret=023f30a320a5781e8ffd1af9888012af incominglimit=10 IP (tos 0x0, ttl 64, id 9242, offset 0, flags [none], proto UDP (17), length 555) 10.0.1.3.5060 10.0.138.226.5060: SIP, length: 527 OPTIONS sip:10.0.138.226 SIP/2.0 Via: SIP/2.0/UDP 10.0.83.61:5060;branch=z9hG4bK6abb74e3;rport Max-Forwards: 70 From: asterisk sip:asterisk@10.0.83.61;tag=as7444eb08 To: sip:10.0.138.226 Contact: sip:asterisk@10.0.83.61:5060 Call-ID: 20afd7e40fb31362355eae245dae1fd6@10.0.83.61:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.2.3 Date: Tue, 29 Mar 2011 17:43:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 IP (tos 0xb8, ttl 250, id 0, offset 0, flags [none], proto UDP (17), length 411) 10.0.138.226.5060 10.0.1.3.5060: SIP, length: 383 SIP/2.0 403 From: URI not recognized Via: SIP/2.0/UDP 10.245.83.61:5060;received=10.0.83.61;branch=z9hG4bK6abb74e3;rport=5060 From: asterisk sip:asterisk@10.0.83.61;tag=as7444eb08 To: sip:10.0.138.226;tag=metaswitch+1+0+e288612a Call-ID: 20afd7e40fb31362355eae245dae1fd6@10.0.83.61:5060 CSeq: 102 OPTIONS Server: DC-SIP/2.0 Organization: Content-Length: 0 IIRC, you need to define the fromuser in the peer in order for the qualify checks (options packets) to contain the user you want -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR MYSQL missing field data
On Tue, Mar 29, 2011 at 12:32 PM, Eric W. Davenport ewdavenp...@certin.comwrote: I restarted Asterisk and I still have all the fields outgoing and incoming except for the CLID field. Clid is populated in the CSV Simple file as well as the CSV Custom file. Please, if you would, copy and paste the output of a 'describe' of your cdr table (in my environment this would be 'describe asterisk.cdr' from the mysql shell). Then, copy and paste the output of 'cat /etc/asterisk/cdr_mysql.conf'. Delete any passwords that file may contain before posting, of course. This will enable us to better troubleshoot your exact issue. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
On Tue, 29 Mar 2011 12:34:04 -0500, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: Remember guys, there's a LOT of IP blocks out there that are almost definitely not going to be somewhere you expect to receive SIP traffic from. On Tue, 29 Mar 2011, Gilles wrote: I agree. Is there a list I could use to check which blocks have been allocated to which countries so I can add them to Asterisk's blacklist? I posted this several months ago: http://www.voip-info.org/wiki/view/allocated-class-a-ip-address-blocks -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wrong from URI in options message
On 3/29/2011 1:56 PM, Sherwood McGowan wrote: [mypeer](peer) host=10.0.138.226 defaultuser=211941 fromuser=211941 md5secret=023f30a320a5781e8ffd1af9888012af incominglimit=10 IIRC, you need to define the fromuser in the peer in order for the qualify checks (options packets) to contain the user you want uhm, didn't I ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wrong from URI in options message
On Tue, Mar 29, 2011 at 1:25 PM, Jeremy Kister asterisk...@jeremykister.com wrote: On 3/29/2011 1:56 PM, Sherwood McGowan wrote: [mypeer](peer) host=10.0.138.226 defaultuser=211941 fromuser=211941 md5secret=023f30a320a5781e8ffd1af9888012af incominglimit=10 IIRC, you need to define the fromuser in the peer in order for the qualify checks (options packets) to contain the user you want uhm, didn't I ? It looks like you did to me. Is it just OPTIONS packets that are showing the wrong fromuser field? In other words, when you send call traffic over this peer, does it properly create the SIP packets? For some reason, I'm thinking this is just the way it is, but someone closer to the the actual sip development may be able to better tell you. Perhaps open a ticket on the bug tracker? -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wrong from URI in options message
On 3/29/2011 2:29 PM, Warren Selby wrote: It looks like you did to me. Is it just OPTIONS packets that are showing the wrong fromuser field? In other words, when you send call traffic over this peer, does it properly create the SIP packets? For some reason, I'm correct - when i actually invite a call or do the register, the from uri is correct. it's just the options packet that is broken. sip development may be able to better tell you. Perhaps open a ticket on the bug tracker? yep, that was the next step - just wanted to run it by a few more eyes before i bothered the devs. https://issues.asterisk.org/view.php?id=19036 -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wrong from URI in options message
Oh, damn, my bad, I've apparently read too many sip.conf entries today -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
Le 29/03/2011 19:34, Sherwood McGowan a écrit : On 3/29/2011 12:25 PM, Steve Edwards wrote: On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan First thing I'd do is restrict the ip blocks your sip endpoints can register/call from in sip.conf (or your database's table for sip endpoints) On Tue, 29 Mar 2011, Gilles wrote: Thanks for the idea, but it's not possible, as the Asterisk must be accessible for road warriors and receive SIP calls from anyone. Really? How many callers are you expecting from North Korea, Libya, China, Iran, etc? Thanks Steve, you just emailed exactly what I was going to say... Remember guys, there's a LOT of IP blocks out there that are almost definitely not going to be somewhere you expect to receive SIP traffic from. Well, I can tell you that our servers in europe those days are mainly attacked by US IP ranges (remember last year the problem with amazon cloud). They now disappear here in europe but lots of other US networks quickly replace them :-( -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
Obviously, the other side of the world wants connections to your side, no matter what side you are on. :-) Cary -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Tuesday, March 29, 2011 3:21 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk and fail2ban Le 29/03/2011 19:34, Sherwood McGowan a écrit : On 3/29/2011 12:25 PM, Steve Edwards wrote: On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan First thing I'd do is restrict the ip blocks your sip endpoints can register/call from in sip.conf (or your database's table for sip endpoints) On Tue, 29 Mar 2011, Gilles wrote: Thanks for the idea, but it's not possible, as the Asterisk must be accessible for road warriors and receive SIP calls from anyone. Really? How many callers are you expecting from North Korea, Libya, China, Iran, etc? Thanks Steve, you just emailed exactly what I was going to say... Remember guys, there's a LOT of IP blocks out there that are almost definitely not going to be somewhere you expect to receive SIP traffic from. Well, I can tell you that our servers in europe those days are mainly attacked by US IP ranges (remember last year the problem with amazon cloud). They now disappear here in europe but lots of other US networks quickly replace them :-( -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
On Tue, Mar 29, 2011 at 3:57 PM, Cary Fitch ca...@usawide.net wrote: Obviously, the other side of the world wants connections to your side, no matter what side you are on. :-) Cary Exactly -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR MYSQL missing field data
On Tue, Mar 29, 2011 at 3:52 PM, Eric W. Davenport ewdavenp...@certin.comwrote: Hi Warren, Thanks for your help, I think this is what you want Please don't mail me (or anyone else) off the list directly without being specifically asked to. The idea is to keep things on the mailing list in order to help other people who may have the same issues. That being said... That information does indeed look like what I want and it appears to be setup correctly. I will be building a comparable test system later today (using all the same software versions as you) and I'll test to see if I get the same issue. If I do, I'll open a new bug on the issue tracker, and I'll post my results here on the list. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
On 03-29-2011 19:25, Steve Edwards wrote: Really? How many callers are you expecting from North Korea, Libya, China, Iran, etc? after reviewing last week's log i'd say around 25-28k/min :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
On Tue, 29 Mar 2011 23:09:06 +0200, ad...@3a.hu wrote: On 03-29-2011 19:25, Steve Edwards wrote: Really? How many callers are you expecting from North Korea, Libya, China, Iran, etc? after reviewing last week's log i'd say around 25-28k/min :) So it looks like I should check out sshguard instead of relying on blocks of IP's :-) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] E1 PRI configuration: DAHDI and LIBPRI
Hi All; I have an E1 card with two ports for ISDN PRI. Do I need to install DAHDI in addition to LIBPRI? For placing outside calls (outgoing) via the PRI, then in the extension.exe file, I will use the Dial function? But how can I determine that I need to use the PRI channels and not the analoge channels? Last point: how can I know that asterisk is containing libpri? In other words, how can I know that the libpri has been included in the compilation and yes I can use it? Any command of file that give this proof? What is the difference if I did not compiled the libpri and if I compiled it? Is there a command to distinguish this? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco IP Phones and Asterisk
Hello; I need to use Cisco IP Phones with Asterisk and I have some questions to know how to use it if someone can advise: 1) How I can assign for each button an extension? 2) How I can assign for specific button a feature to be used (like call forward or call pickup .. etc)? 3) As you know that it is required to have a correct username and password to login, so where to give the username and password in the Cisco IP Phone to be able to login for the SIP account? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
On 03-29-2011 19:25, Steve Edwards wrote: Really? How many callers are you expecting from North Korea, Libya, China, Iran, etc? On Tue, 29 Mar 2011 23:09:06 +0200, ad...@3a.hu wrote: after reviewing last week's log i'd say around 25-28k/min :) On Tue, 29 Mar 2011, Gilles wrote: So it looks like I should check out sshguard instead of relying on blocks of IP's :-) It's not A or B, think A AND B. Security should be in layers -- my pocket GPS is in my locked glove box, in my locked car, in my locked garage, in my gated community. If there is never a need to accept callers from North Korea, how will you explain to your boss that some NK script weenie discovered some weakness in A or B and racked up a bazillion minutes to Libya? What if you misconfigure A or B? What if A or B has a 'window of opportunity' during system restart? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
Hi Gilles, Just to provide an alternative to sshguard: you could use BFD[1] (based on bash scripts) and configure it to use iptables to block the attacker host. The default configuration is to check the logs at each 3 minutes (using a crontab entry). BFD rules for Asterisk could be found here [2] - tested on Asterisk 1.4 Our BAN command looks like: (/sbin/iptables -n -L | grep DROP | grep $ATTACK_HOST) || /sbin/ipttables -I INPUT -s $ATTACK_HOST -j DROP HTH, Ioan [1] http://www.rfxn.com/projects/brute-force-detection/ [2] http://www.modulo.ro/Modulo/downloads/tools/tenora.bfd.tar.gz On Wed, Mar 30, 2011 at 12:51 AM, Gilles codecompl...@free.fr wrote: On Tue, 29 Mar 2011 23:09:06 +0200, ad...@3a.hu wrote: On 03-29-2011 19:25, Steve Edwards wrote: Really? How many callers are you expecting from North Korea, Libya, China, Iran, etc? after reviewing last week's log i'd say around 25-28k/min :) So it looks like I should check out sshguard instead of relying on blocks of IP's :-) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1 PRI configuration: DAHDI and LIBPRI
Sorry for the top post, responding from my phone... Yes, you'll need both DAHDI and libpri to make an E1 card work with asterisk. Yes, you'll most likely use the Dial() command inside extensions.conf in order to dial out. You'll differentiate your PRI channels from your analog channels in your chan_dahdi.conf file (I believe), probably using groups (although I suppose you could just use channel numbers). I believe you can verify that libpri has been compiled into asterisk by issuing a 'pri show' command at the console after you've got everything running. Thanks, --Warren Selby, dCAP On Mar 29, 2011, at 5:11 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; I have an E1 card with two ports for ISDN PRI. Do I need to install DAHDI in addition to LIBPRI? For placing outside calls (outgoing) via the PRI, then in the extension.exe file, I will use the Dial function? But how can I determine that I need to use the PRI channels and not the analoge channels? Last point: how can I know that asterisk is containing libpri? In other words, how can I know that the libpri has been included in the compilation and yes I can use it? Any command of file that give this proof? What is the difference if I did not compiled the libpri and if I compiled it? Is there a command to distinguish this? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Asterisk
The answer to all of your questions are the same - the config file that you create for your phone. Thanks, --Warren Selby, dCAP On Mar 29, 2011, at 5:16 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hello; I need to use Cisco IP Phones with Asterisk and I have some questions to know how to use it if someone can advise: 1) How I can assign for each button an extension? 2) How I can assign for specific button a feature to be used (like call forward or call pickup .. etc)? 3) As you know that it is required to have a correct username and password to login, so where to give the username and password in the Cisco IP Phone to be able to login for the SIP account? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR MYSQL missing field data
On Tue, Mar 29, 2011 at 4:03 PM, Warren Selby wcse...@selbytech.com wrote: That information does indeed look like what I want and it appears to be setup correctly. I will be building a comparable test system later today (using all the same software versions as you) and I'll test to see if I get the same issue. If I do, I'll open a new bug on the issue tracker, and I'll post my results here on the list. Well, after recreating your environment in a virtual machine tonight, I was able to reproduce the issue, so I created a bug on the issue tracker: https://issues.asterisk.org/view.php?id=19040 Hopefully someone a little closer to the development will be able to take a look and see what's going on. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR MYSQL missing field data
I listened to your email using DriveCarefully and will respond as soon as I can. Download DriveCarefully for free at www.drivecarefully.com -- Sent from my BlackBerry® Senior Support Engineer US Numbers: 561-886-0664 Nicaragua Mobile: +505.8488.6876 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Discover when remote phone answers through IAX2
Hi, I'm using IAX2 between our SIP and PSTN servers, both running Asterisk 1.6.2. Users connect to the SIP server and dial; the SIP server forwards the call to the PSTN server over IAX2, which then dials out over the connected PRI. Since users need detailed call progress feedback, the first action in the dialplan on the PSTN server side is Answer(). In this scenario it's easy for a human to know when a call has been answered. However, the SIP-side Asterisk treats the call as answered the moment the PSTN server executes Answer(). Is there any way of determining on the SIP side when the called party actually picks up the phone? Or if she doesn't, the status of the call as it progresses? Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
