Re: [asterisk-users] standalone PRI-to-SIP converter
On 5/27/11 6:33 PM, Gordon Henderson wrote: Personally I'd avoid Patton. No-one has a clue how to configure them. I've struggled for the past couple of days and have given up and they're being sent back to be replaced by Mediatrix boxes. Then you're asking the wrong people. It is totally possible to get a Patton to be configured correctly. Since PRI is much easier to configure than a BRI interface (PtP, PtMP?) it shouldn't be that hard. The problem with these very powerful VoIP to ISDN gateways is that they have lots of things to configure, some more intuitive than others. If you're using real hardware, be prepared to spend real time and effort into configuring them. The webinterfaces on Patton or Audiocodes gateways are miles better than the CLI on a Cisco AS5350 or the CLI on an Acme Packet SBC. The bad rep Patton and Audiocodes seem to have is probably related to them using the same software for a simple 2xFXO port gateway as those for 4xISDN BRI or 4x ISDN PRI. -- Andreas Sikkema -- _ -- 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] Does anyone know about asterisk 1.10
I read about asterisk 1.10 in website https://wiki.asterisk.org. but didnt find this release from asterisk community. -- _ -- 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] Question about null routing calls to DIDs we don't handle
On 1 Jun 2011, at 22:50, Jesse Thompson wrote:
We are managing an Asterisk installation for residential VOIP service, and we
are having a problem where all inbound calls to DIDs which are assigned to us
by our wholesaler but not yet assigned to a downstream customer get caught in
a routing loop.
Put this line:
_NXXNXX = Dial(SIP/${EXTEN}@upstream,120); // numbers not handled here get
routed upstream
in the 'local' context instead of the other one? Letting a carrier use you
as a carrier seems like quite a bad idea generally..
S
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Re: [asterisk-users] Does anyone know about asterisk 1.10
See this link for release date... https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions [SATISH] On Thu, Jun 2, 2011 at 1:09 PM, Nikhil d.nik...@cem-solutions.net wrote: I read about asterisk 1.10 in website https://wiki.asterisk.org. but didnt find this release from asterisk community. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] please help
OK thanks a lot for your help now all is ok :)
2011/5/31 salaheddine elharit salah.elharit...@gmail.com
Hello
after remove the _ and put the number like that 0678922645,1, the issue has
been solved
thank you so much :)
2011/5/31 mahesh katta maheshka...@flexydial.com
Remove the _ in front of your dialplan,like
exten = 0678922645,1,--
On Mon, May 30, 2011 at 11:00 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
Hello list
i have configured astersik 1.4 with sip i have a question
when i put in dial plan.conf
exten = _0678922645.,1,Set(CALLERID(number)=520460587)
exten = _0678922645
.,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = _0678922645
.,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded))
exten = _067892264*5*,2,Hangup()
i can not call my number but when i delet the last number '5' i can call
without any issue
i want to put all the number please any hel to solve this issue
thanks and regards
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Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
(E) Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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Re: [asterisk-users] Three-way conference in Asterisk
Nikhil,
This is how I would implement '3 way conference' in Asterisk with the help
of dynamic features.
Assume 3 SIP friends 1110, and 1112 in sip.conf. For 1110 in sip.conf,
context=test3way
Add following in applicationmap section of features.conf
[applicationmap]
3way-start = **0,caller,Macro,3way-start
3way-conf = **1,caller,Macro,3way-conf
3way-noconf = **2,caller,Macro,3way-noconf
My dialplan would be
[test3way]
exten = 1212,1,Noop(## TLC Check ##)
same = n,set(DYNAMIC_FEATURES=3way-start)
same = n,Dial(SIP/,30,m)
[dynamic-3way]
exten = _XXX.,1,Answer
exten = _XXX.,n,Set(CONFNO=1212)
exten = _XXX.,n,Set(DYNAMIC_FEATURES=)
exten = _XXX.,n,ConfBridge(${CONFNO},M)
exten = _XXX.,n,Hangup
[macro-3way-start]
exten = s,1,Set(CONFNO=1212)
exten = s,n,ChannelRedirect(${BRIDGEPEER},dynamic-3way,${CONFNO},1)
exten = s,n,wait(1)
exten = s,n,Set(DYNAMIC_FEATURES=3way-conf#3way-noconf)
exten = s,n,Dial(SIP/1112,,g)
exten = s,n,Set(DYNAMIC_FEATURES=)
exten = s,n,ConfBridge(${CONFNO},M)
[macro-3way-conf]
exten = s,1,ChannelRedirect(${BRIDGEPEER},dynamic-3way,${CONFNO},1)
[macro-3way-noconf]
exten = s,1,SoftHangup(${BRIDGEPEER})
You can dial 1212 from SIP Extension 1110 which will connect 1110 to .
No while talking to , 1110 can press **0 to invoke '3way-start' feature
which in turn call 1112.
Now while talking to 1112, 1110 can press **1 to start the conference.
I suggest you should go through features.conf for more information. This is
very basic dialplan for 3 way conference. You will have to add some more
stuffs to make it work in the way you want.
Hope this helps.
[SATISH]
On Thu, Jun 2, 2011 at 11:25 AM, Nikhil d.nik...@cem-solutions.net wrote:
Hi
How to set a threeway conference in asterisk only for VOIP (I am using
only SIP channel).
Thanks
Nikhil
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Re: [asterisk-users] standalone PRI-to-SIP converter
Hi everyone, i ended using Patton, right after a fiasco with overlap to enblock conversion on a users PBX ... Had to go out and pay more for a Voxip unit . That things serves me well and is far less complicated to setup. NOw using it to interconnect a PBX with aprox 5000 extensions behind it, without a problem. But ... on the side note . MAKE THEM CONVERT TO IP PBXs .. ( though hardly possible on large installs ) M S, Andreas Sikkema piše: On 5/27/11 6:33 PM, Gordon Henderson wrote: Personally I'd avoid Patton. No-one has a clue how to configure them. I've struggled for the past couple of days and have given up and they're being sent back to be replaced by Mediatrix boxes. Then you're asking the wrong people. It is totally possible to get a Patton to be configured correctly. Since PRI is much easier to configure than a BRI interface (PtP, PtMP?) it shouldn't be that hard. The problem with these very powerful VoIP to ISDN gateways is that they have lots of things to configure, some more intuitive than others. If you're using real hardware, be prepared to spend real time and effort into configuring them. The webinterfaces on Patton or Audiocodes gateways are miles better than the CLI on a Cisco AS5350 or the CLI on an Acme Packet SBC. The bad rep Patton and Audiocodes seem to have is probably related to them using the same software for a simple 2xFXO port gateway as those for 4xISDN BRI or 4x ISDN PRI. -- _ -- 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] Migration from Mantis to JIRA
We use Jira at work. I hate it. Hope you have a better experience than I've had! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russell Bryant Sent: Wednesday, June 01, 2011 7:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Migration from Mantis to JIRA Greetings, A few weeks ago I posted a message about the upcoming migration from Mantis to JIRA for issues.asterisk.org [1]. A lot of testing has been done and all known issues have been resolved. We have scheduled the migration for Sunday, June 5th. The issue tracker will be down most of the day as the migration takes place. Once the migration is complete, the issue tracker will be: https://issues.asterisk.org/jira/ Mantis will still be available for some time, but will be read-only. If you have an account on Mantis, you will be able to log in to JIRA using the same username. All of your history will have been migrated. This account can also be used on wiki.asterisk.org. IMPORTANT NOTE: You will have to click the forgot my password link to reset your password before you can log in, though. It is not possible to migrate passwords from one to the other as they use a different hashing algorithm. For more information about how to use JIRA, see the JIRA user's guide: http://confluence.atlassian.com/display/JIRA042/JIRA+User%27s+Guide If you run into any problems after the migration has taken place, please report them in the JIRA Help project. If you would rather report something via email, email espiceland at digium dot com and me. Thanks, [1] http://lists.digium.com/pipermail/asterisk-dev/2011-May/049088.html -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] benefits of asterisk 1.8
can someone explain to me the benefits of upgrading to version 1.8? we are currently running 1.6 I know one benefit of 1.8 is digium supports it also, how stable is version 1.8 compared to 1.6? Thank you for you input. -- _ -- 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] benefits of asterisk 1.8
1.8 is stable when compared to 1.6, also in 1.8 you will get Long Term support. On Thu, Jun 2, 2011 at 6:31 PM, vip killa vipki...@gmail.com wrote: can someone explain to me the benefits of upgrading to version 1.8? we are currently running 1.6 I know one benefit of 1.8 is digium supports it also, how stable is version 1.8 compared to 1.6? Thank you for you input. -- _ -- 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] benefits of asterisk 1.8
So many new features have been added in 1.8. Check this...https://wiki.asterisk.org/wiki/display/AST/New+in+1.8 Nope, Asterisk 1.8 is not stable enough yet. [SATISH] On Thu, Jun 2, 2011 at 6:33 PM, Gopal krishnan gopalakrishnan...@gmail.comwrote: 1.8 is stable when compared to 1.6, also in 1.8 you will get Long Term support. On Thu, Jun 2, 2011 at 6:31 PM, vip killa vipki...@gmail.com wrote: can someone explain to me the benefits of upgrading to version 1.8? we are currently running 1.6 I know one benefit of 1.8 is digium supports it also, how stable is version 1.8 compared to 1.6? Thank you for you input. -- _ -- 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 -- _ -- 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] benefits of asterisk 1.8
what do you mean Asterisk 1.8 is not stable enough yet? Can you give specific examples/scenarios? On Thu, Jun 2, 2011 at 9:28 AM, Satish Barot satish4aster...@gmail.comwrote: So many new features have been added in 1.8. Check this...https://wiki.asterisk.org/wiki/display/AST/New+in+1.8 Nope, Asterisk 1.8 is not stable enough yet. [SATISH] On Thu, Jun 2, 2011 at 6:33 PM, Gopal krishnan gopalakrishnan.an@ gmail.com wrote: 1.8 is stable when compared to 1.6, also in 1.8 you will get Long Term support. On Thu, Jun 2, 2011 at 6:31 PM, vip killa vipki...@gmail.com wrote: can someone explain to me the benefits of upgrading to version 1.8? we are currently running 1.6 I know one benefit of 1.8 is digium supports it also, how stable is version 1.8 compared to 1.6? Thank you for you input. -- _ -- 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 -- _ -- 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] RealTime Queue Logging in 1.8
Hi Does anyone know of an accurate resource I could refer to for this? The best I can find is http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL And that table wont create in my database... Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Playing with sipvicious ..
On 01/06/11 16:13, Allen David Niven wrote: what does ossec give u that fail2ban does not ? thx and cheers Replied to list so others can find this in the future if they want to. I haven't spent a lot of time investigating fail2ban as I was already using ossec before I saw much talk about fail2ban with Asterisk. Anyway as far as I can see my main advantage is that OSSEC has multiple levels of incidents. So I can create rules to send emails out for unusual activity that might not necessarily require an IP block but needs checking out. My fear with something that just watches Asterisk logs for a very specific known attack metric and then blocks IP(s) based on that is what happens when the attackers start doing something different? Fail2ban may well do all this as well, I don't know but I find OSSEC does it very well and the XML rules and log decoders are very versatile. cheers, Paul. -- _ -- 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] AMI buffering event output?
To clarify; I observe the exact same results no matter how I connect to the AMI on this particular server. I tried connecting FROM this server to an AMI on another server to make sure it wasn't the telnet client or some such, and then it worked perfectly. To answer the question, if I use the external IP address rather than 127.0.0.1 I observe the same results. -- Örn On Thu, Jun 2, 2011 at 3:19 AM, Matt Riddell li...@venturevoip.com wrote: On 1/06/11 11:03 PM, Örn Arnarson wrote: Hi Matt, Yes, passing two carriage returns. I login successfully. Here's example output (with my comments in []) Trying 127.0.0.1... Connected to localhost. Escape character is '^]'. Asterisk Call Manager/1.1 action: login username: phpagi secret: supersecretpassword events: on Response: Success Message: Authentication accepted It seems somewhat impossible that you would be getting different results from different hosts. Are you using the same login? What if you use the external IP rather than 127.0.0.1 -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] benefits of asterisk 1.8
On 11-06-02 09:35 AM, vip killa wrote: what do you mean Asterisk 1.8 is not stable enough yet? Can you give specific examples/scenarios? I too would like to see a specific example, additionally if you can create an test using the testsuite I'll be happy to review it and merge the code into subversion. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] benefits of asterisk 1.8
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Thursday, June 02, 2011 11:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] benefits of asterisk 1.8 On 11-06-02 09:35 AM, vip killa wrote: what do you mean Asterisk 1.8 is not stable enough yet? Can you give specific examples/scenarios? I too would like to see a specific example, additionally if you can create an test using the testsuite I'll be happy to review it and merge the code into subversion. Does Digium run 1.8 on their production corporate PBX? -- _ -- 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] AMI buffering event output?
2011/6/2 Örn Arnarson o...@arnarson.net: To clarify; I observe the exact same results no matter how I connect to the AMI on this particular server. I tried connecting FROM this server to an AMI on another server to make sure it wasn't the telnet client or some such, and then it worked perfectly. To answer the question, if I use the external IP address rather than 127.0.0.1 I observe the same results. echo $LANG on each server ? -- _ -- 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] Migration from Mantis to JIRA
On 06/02/2011 06:46 AM, Terry Brummell wrote: We use Jira at work. I hate it. Hope you have a better experience than I've had! We've been using it for years internally to Digium. We've been happy with it. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] benefits of asterisk 1.8
On 06/02/2011 10:29 AM, Eric Wieling wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Thursday, June 02, 2011 11:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] benefits of asterisk 1.8 On 11-06-02 09:35 AM, vip killa wrote: what do you mean Asterisk 1.8 is not stable enough yet? Can you give specific examples/scenarios? I too would like to see a specific example, additionally if you can create an test using the testsuite I'll be happy to review it and merge the code into subversion. Does Digium run 1.8 on their production corporate PBX? We have two Asterisk systems that comprise our PBX: one is a Switchvox system that handles the bulk of the duties, and there is an Asterisk 1.8 system connected to it that handles all of the stuff Switchvox isn't really designed for. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] standalone PRI-to-SIP converter
On Thu, 2 Jun 2011, Andreas Sikkema wrote: On 5/27/11 6:33 PM, Gordon Henderson wrote: Personally I'd avoid Patton. No-one has a clue how to configure them. I've struggled for the past couple of days and have given up and they're being sent back to be replaced by Mediatrix boxes. Then you're asking the wrong people. It is totally possible to get a Patton to be configured correctly. Since PRI is much easier to configure than a BRI interface (PtP, PtMP?) it shouldn't be that hard. Fine, but I've asked here twice about Patton units - had one reply back from someone who'd been helpful, but when times pressing, it's easier to just use something you've used before. Gordon -- _ -- 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] Playing with sipvicious ..
Well,
About sipvicious, just put a kamailio in front of asterisk and just drop
all messages with user agents corrreponding to these messages.
Spivicious first send options messages, read the user agent and drop if
it's corresponding to one of the user agents well known to be used.
In Kamailio (to be updtaed) I have :
### Country check
if (is_method(OPTIONS) || is_method(REGISTER))
{
avp_db_query(SELECT sql_cache country FROM ip_country inner
join GeoLiteCity on GeoLiteCity.locId = ip_country.locId WHERE
MBRCONTAINS(ip_poly, POINTFROMWKB(POINT(INET_ATON('$si'), 0))) limit 1;
,$avp(s:countryCode));
if ($avp(s:countryCode) !=BE $avp(s:countryCode) !=FR
$avp(s:countryCode) !=LU $avp(s:countryCode) !=MA
$avp(s:countryCode) !=ES $avp(s:countryCode) !=IT
$avp(s:countryCode) !=DE )
{
xlog(L_NOTICE, -- Probable Attack
attempt from countrycode : $avp(s:countryCode) - $si - $fu - $ua - $rm );
}
}
### Hackers check
if($ua==friendly-scanner){
xlog(L_NOTICE, -- Attack attempt from
countrycode : $avp(s:countryCode) - $si - $fu - $ua - $rm - DROP );
drop();
}
if($ua==sundayddr){
xlog(L_NOTICE, -- Attack attempt from
countrycode : $avp(s:countryCode) - $si - $fu - $ua - $rm - DROP );
drop();
}
if($ua==sip-scan){
xlog(L_NOTICE, -- Attack attempt from
countrycode : $avp(s:countryCode) - $si - $fu - $ua - $rm - DROP );
drop();
}
if($ua==iWar){
xlog(L_NOTICE, -- Attack attempt from
countrycode : $avp(s:countryCode) - $si - $fu - $ua - $rm - DROP );
drop();
}
if($ua==sipsak){
xlog(L_NOTICE, -- Attack attempt from
countrycode : $avp(s:countryCode) - $si - $fu - $ua - $rm - DROP );
drop();
}
When sipvicious doesn't receive answer, it stops scanning the server :)
Best regards,
Olivier
Le 2/06/11 17:06, Paul Hayes a écrit :
On 01/06/11 16:13, Allen David Niven wrote:
what does ossec give u that fail2ban does not ?
thx and cheers
Replied to list so others can find this in the future if they want to.
I haven't spent a lot of time investigating fail2ban as I was already
using ossec before I saw much talk about fail2ban with Asterisk.
Anyway as far as I can see my main advantage is that OSSEC has
multiple levels of incidents. So I can create rules to send emails
out for unusual activity that might not necessarily require an IP
block but needs checking out.
My fear with something that just watches Asterisk logs for a very
specific known attack metric and then blocks IP(s) based on that is
what happens when the attackers start doing something different?
Fail2ban may well do all this as well, I don't know but I find OSSEC
does it very well and the XML rules and log decoders are very versatile.
cheers,
Paul.
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Re: [asterisk-users] SIP Register DOS attack
I'll check this option and see if it helps next time, just to clarify, there were no actual calls in place, just DOS register attack. On Wed, Jun 1, 2011 at 12:22 PM, Ira i...@extrasensory.com wrote: At 10:56 AM 6/1/2011, you wrote: Do you have: sip.conf [general] allowguest=no So because of this I decided to type sip show channels into my Asterisk and got this: Peer User/ANRCall ID Format Hold Last Message Expiry Peer 216.xxx.69.xxx (None) f2d8db55-0a7edd (nothing) NoRx: OPTIONS guest 216.xxx.69.xxx (None) 2ce0b9a5-6de7f4 (nothing) NoRx: OPTIONS guest 64.xxx.41.xxx6314098389 2a482e4b684a59a (nothing) No guest 192.168.233.xxx (None) ioh3fna2aw.n4mz (nothing) NoRx: REGISTER guest 4 active SIP dialogs I have allowguest=no and all of those IPs are either my providers or a SIP phone on my network so why would it show guest as the peer? I'm running Asterisk SVN-trunk-r319759M if that matters. Ira -- _ -- 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] Free CNAM
If you can use curl, and can do some text parsing and know regular expressions, you may be able to use this free CNAM service: http://www.numberguru.com/ and integrate into your system. This one appears to have a more complete database. When I tried my number, I have gotten my full name, but when I use the FreeCNAM project below, I just get Florida. On Wed, Jun 1, 2011 at 8:11 AM, Michael R. Wally michael.r.wa...@gmail.comwrote: I've been toying around with the idea of starting some kind of 'Open CNAM' project to destroy the current money hustle BS that dominates this industry. The ever-growing FreeCNAM database may be a good starting point for such a project. I would also like to use Bitcoin (BTC) as the micropayment solution for user-requested updates. Some nominal fee. If anyone wants to get involved, contact me. On 06/01/2011 07:51 AM, Skyler wrote: Hi, The junk in CNAM databases like FLORIDA, ONTARIO etc. is IMO the carrier's way to isolate their users and another excuse to charge more money for 'the better plan'. In the end, it's the carrier that inputs the info so if it shows FLORIDA with one database I can't see how any other database would be different as the carrier is the only one that controls the outbound CID info. Calling me from POTS to snatch the CID will result in the same. ...unless there were a user friendly CNAM service, where info could be updated by the end-user and queried freely by voip providers. I would update my cellular numbers for sure and know at least a dozen people that would do the same. Everyone is going VoIP so why not? Talking about 'where's the money or angle'... here is one, vanity. Charge $1/yr to a user per DID, if I don't renew then delete it and re-query the original carrier. 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 -- Pascal B. Personal Web Site http://www.pascalbruno.com/ Twitter: @petchaw -- _ -- 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] Can I use phone line to recive faxes?
Hi Guys, Actually My question is as in the subject, may I use a regular phone line to receive faxes with FFA (Fax For Asterisk), I am using asterisk 1.6.2.8. -- Khalid -- _ -- 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] Free CNAM
I just checked several of my numbers and several others known to me, it really isn't much better 2 of them returned names other than mine, and all had the wrong city, though at least the state was correct. All but one also had the wrong carrier. I fear these databases are are so full of errors that they are mostly worthless. John Novack Pascal Bruno wrote: If you can use curl, and can do some text parsing and know regular expressions, you may be able to use this free CNAM service: http://www.numberguru.com/ and integrate into your system. This one appears to have a more complete database. When I tried my number, I have gotten my full name, but when I use the FreeCNAM project below, I just get Florida. On Wed, Jun 1, 2011 at 8:11 AM, Michael R. Wally michael.r.wa...@gmail.com mailto:michael.r.wa...@gmail.com wrote: I've been toying around with the idea of starting some kind of 'Open CNAM' project to destroy the current money hustle BS that dominates this industry. The ever-growing FreeCNAM database may be a good starting point for such a project. I would also like to use Bitcoin (BTC) as the micropayment solution for user-requested updates. Some nominal fee. If anyone wants to get involved, contact me. On 06/01/2011 07:51 AM, Skyler wrote: Hi, The junk in CNAM databases like FLORIDA, ONTARIO etc. is IMO the carrier's way to isolate their users and another excuse to charge more money for 'the better plan'. In the end, it's the carrier that inputs the info so if it shows FLORIDA with one database I can't see how any other database would be different as the carrier is the only one that controls the outbound CID info. Calling me from POTS to snatch the CID will result in the same. ...unless there were a user friendly CNAM service, where info could be updated by the end-user and queried freely by voip providers. I would update my cellular numbers for sure and know at least a dozen people that would do the same. Everyone is going VoIP so why not? Talking about 'where's the money or angle'... here is one, vanity. Charge $1/yr to a user per DID, if I don't renew then delete it and re-query the original carrier. 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 -- Pascal B. Personal Web Site http://www.pascalbruno.com/ Twitter: @petchaw -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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] SIP Register DOS attack
Also you guys may need to use: sip.conf [general] allowguest=no *alwaysauthreject = yes* On Thu, Jun 2, 2011 at 1:01 PM, Al lists asteris...@gmail.com wrote: I'll check this option and see if it helps next time, just to clarify, there were no actual calls in place, just DOS register attack. On Wed, Jun 1, 2011 at 12:22 PM, Ira i...@extrasensory.com wrote: At 10:56 AM 6/1/2011, you wrote: Do you have: sip.conf [general] allowguest=no So because of this I decided to type sip show channels into my Asterisk and got this: Peer User/ANRCall ID Format Hold Last Message Expiry Peer 216.xxx.69.xxx (None) f2d8db55-0a7edd (nothing) NoRx: OPTIONS guest 216.xxx.69.xxx (None) 2ce0b9a5-6de7f4 (nothing) NoRx: OPTIONS guest 64.xxx.41.xxx6314098389 2a482e4b684a59a (nothing) No guest 192.168.233.xxx (None) ioh3fna2aw.n4mz (nothing) NoRx: REGISTER guest 4 active SIP dialogs I have allowguest=no and all of those IPs are either my providers or a SIP phone on my network so why would it show guest as the peer? I'm running Asterisk SVN-trunk-r319759M if that matters. Ira -- _ -- 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 -- Abdullah -- _ -- 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] How to continue processing a context after a Hangup
Good afternoon, I'm trying to write a simple callback context, but i need to hangup an incoming call and then call the origin number back, the problem is that asterisk stops processing the call after Hangup() application then it is not able to dial the origin number back. Sorry for the grammatical erros. -- _ -- 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 1.8.4.2 Now Available (Security Release)
The Asterisk Development Team has announced the release of Asterisk version 1.8.4.2, which is a security release for Asterisk 1.8. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of Asterisk 1.8.4.2 resolves an issue with SIP URI parsing which can lead to a remotely exploitable crash: Remote Crash Vulnerability in SIP channel driver (AST-2011-007) The issue and resolution is described in the AST-2011-007 security advisory. For more information about the details of this vulnerability, please read the security advisory AST-2011-007, which was released at the same time as this announcement. For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.2 Security advisory AST-2011-007 is available at: http://downloads.asterisk.org/pub/security/AST-2011-007.pdf Thank you for your continued support of Asterisk! -- _ -- 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] AMI buffering event output?
en_US.UTF-8 in all cases. On Thu, Jun 2, 2011 at 3:33 PM, Mark Deneen mden...@gmail.com wrote: 2011/6/2 Örn Arnarson o...@arnarson.net: To clarify; I observe the exact same results no matter how I connect to the AMI on this particular server. I tried connecting FROM this server to an AMI on another server to make sure it wasn't the telnet client or some such, and then it worked perfectly. To answer the question, if I use the external IP address rather than 127.0.0.1 I observe the same results. echo $LANG on each server ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 83, Issue 3
Letting a carrier use you as a carrier seems like quite a bad idea generally..
I think I would agree. :)
_NXXNXX = Dial(SIP/${EXTEN}@upstream,120); // numbers not handled here
get routed upstream
in the 'local' context instead of the other one?
So here is where the finer points of Asterisk pattern matching must
come into play.
All of the customer DID's match the pattern _NXXNXX. If we put
that pattern in the local context, then wouldn't that mean that calls
from a local customer to another local customer would match the
_NXXNXX pattern before even trying to match against the specific
patterns in the clients context? We need to be able to route
local-to-local calls without using two trunks to go back and forth
through the upstream provider.
Thank you for your input. I know this is a problem most operators can
get past, so there's got to be just something not lining up quite
right in my mental model. :)
- - Jesse
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Re: [asterisk-users] Asterisk 1.8.4.2 Now Available (Security Release)
Is this available in current SVN ? Date: Thu, 2 Jun 2011 15:07:50 -0400 From: asteriskt...@digium.com To: asteriskt...@digium.com Subject: [asterisk-users] Asterisk 1.8.4.2 Now Available (Security Release) The Asterisk Development Team has announced the release of Asterisk version 1.8.4.2, which is a security release for Asterisk 1.8. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The release of Asterisk 1.8.4.2 resolves an issue with SIP URI parsing which can lead to a remotely exploitable crash: Remote Crash Vulnerability in SIP channel driver (AST-2011-007) The issue and resolution is described in the AST-2011-007 security advisory. For more information about the details of this vulnerability, please read the security advisory AST-2011-007, which was released at the same time as this announcement. For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.2 Security advisory AST-2011-007 is available at: http://downloads.asterisk.org/pub/security/AST-2011-007.pdf Thank you for your continued support of Asterisk! -- _ -- 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] asterisk logger permission
Hi Guys! If i reload my asterisk it create /var/log/asterisk/* file with root permission. I am running asterisk with asterisk user and group. Do you have any idea ? root@campbx1:~# ls -l /var/log/asterisk/ total 716 drwxr-xr-x 2 asterisk asterisk 4096 2011-05-06 15:38 cdr-csv drwxr-xr-x 2 asterisk asterisk 4096 2011-03-22 14:53 cdr-custom drwxr-xr-x 2 asterisk asterisk 4096 2011-03-22 14:53 cel-csv drwxr-xr-x 2 asterisk asterisk 4096 2011-03-22 14:53 cel-custom -rw-r- 1 root root 0 2011-05-15 06:25 full -rw-r- 1 asterisk asterisk 617026 2011-05-15 06:25 full.1 -rw-r--r-- 1 asterisk asterisk 41439 2011-05-08 11:24 full.2.gz -rw-r- 1 root root 0 2011-05-15 06:25 messages -rw-r- 1 asterisk asterisk 36519 2011-05-14 19:29 messages.1 -rw-r--r-- 1 asterisk asterisk 2520 2011-05-06 17:21 messages.2.gz -rw-r- 1 root root 0 2011-05-15 06:25 queue_log -rw-r--r-- 1 asterisk asterisk392 2011-05-12 17:23 queue_log.1 -- _ -- 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] Free CNAM
Hi all, Let's get some feedback going here and see if there is any general support in a user-driven CNAM concept. Assuming that your landline/mobile outbound provider does not push caller-name + number for you with your calling plan. Would you pay $1/yr to have the access to update your own personal CNAM info in a database that you can trust to be correct? One that 1000's or even 100,000's of other voip/pbx owners will use? S. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Thursday, June 02, 2011 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Free CNAM I just checked several of my numbers and several others known to me, it really isn't much better 2 of them returned names other than mine, and all had the wrong city, though at least the state was correct. All but one also had the wrong carrier. I fear these databases are are so full of errors that they are mostly worthless. John Novack Pascal Bruno wrote: If you can use curl, and can do some text parsing and know regular expressions, you may be able to use this free CNAM service: http://www.numberguru.com/ and integrate into your system. This one appears to have a more complete database. When I tried my number, I have gotten my full name, but when I use the FreeCNAM project below, I just get Florida. On Wed, Jun 1, 2011 at 8:11 AM, Michael R. Wally michael.r.wa...@gmail.com wrote: I've been toying around with the idea of starting some kind of 'Open CNAM' project to destroy the current money hustle BS that dominates this industry. The ever-growing FreeCNAM database may be a good starting point for such a project. I would also like to use Bitcoin (BTC) as the micropayment solution for user-requested updates. Some nominal fee. If anyone wants to get involved, contact me. On 06/01/2011 07:51 AM, Skyler wrote: Hi, The junk in CNAM databases like FLORIDA, ONTARIO etc. is IMO the carrier's way to isolate their users and another excuse to charge more money for 'the better plan'. In the end, it's the carrier that inputs the info so if it shows FLORIDA with one database I can't see how any other database would be different as the carrier is the only one that controls the outbound CID info. Calling me from POTS to snatch the CID will result in the same. ...unless there were a user friendly CNAM service, where info could be updated by the end-user and queried freely by voip providers. I would update my cellular numbers for sure and know at least a dozen people that would do the same. Everyone is going VoIP so why not? Talking about 'where's the money or angle'... here is one, vanity. Charge $1/yr to a user per DID, if I don't renew then delete it and re-query the original carrier. S. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Pascal B. Personal Web Site http://www.pascalbruno.com/ Twitter: @petchaw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot _ No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1375 / Virus Database: 1511/3675 - Release Date: 06/02/11 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to continue processing a context after a Hangup
2011/6/2 Antonio Modesto mode...@isimples.com.br
Good afternoon,
I'm trying to write a simple callback context, but i need to hangup an
incoming call and then call the origin number back, the problem is that
asterisk stops processing the call after Hangup() application then it is not
able to dial the origin number back.
The way I did it was to use a DeadAGI from the 'h' exten that created a call
file. This is how I did it for a client on Asterisk 1.4.x:
[rec-call-back-in]
exten = new,1,Answer()
exten = new,n,Wait(1)
exten = new,n,Playback(vm-intro)
exten = new,n,Playback(beep)
exten = new,n,Set(timestamp=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten = new,n,Set(FILENAME=reccallback/${CALLERID(num)}-${timestamp})
exten = new,n,Record(${FILENAME}.gsm,,,q)
exten = new,n,Playback(vm-goodbye)
exten = new,n,Hangup()
exten = h,1,Verbose(Hangup after recording)
exten = h,n,DeadAGI(reccallback.agi,${FILENAME},${TIMESTAMP})
[rec-call-back-out]
exten = out,1,Wait(2)
exten = out,n,Playback(${playbackfile})
exten = out,n,Hangup()
reccallback.agi:
#!/usr/bin/perl
use Asterisk::AGI;
use File::Copy;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
my $callerid = $input{'callerid'};
my $recfile = $ARGV[0];
my $timestamp = $ARGV[1];
open CALLFILE, /var/spool/asterisk/tmp/$callerid-$timestamp.call;
if (length($callerid) 4) {
print CALLFILE Channel: SIP/external-sip-provider/+1$callerid\n;
} else {
print CALLFILE Channel: SIP/$callerid\n;
}
print CALLFILE CallerID: \CUSTOMER\ XX\n;
print CALLFILE MaxRetries: 2\n;
print CALLFILE RetryTime: 60\n;
print CALLFILE WaitTime: 20\n;
print CALLFILE Context: rec-call-back-out\n;
print CALLFILE Extension: out\n;
print CALLFILE Priority: 1\n;
print CALLFILE Set: playbackfile=$recfile\n;
close CALLFILE;
sleep(5);
copy(/var/spool/asterisk/tmp/$callerid-$timestamp.call,
/var/spool/asterisk/outgoing/$callerid-$timestamp.call) or die copy
failed: $!;
exit;
--
Thanks,
--Warren Selby, dCAP
Our website just got a facelift! Check it out!
http://www.SelbyTech.com http://www.selbytech.com
--
_
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Re: [asterisk-users] Asterisk 1.8.4.2 Now Available (Security Release)
On 02/06/11 03:35 PM, satish patel wrote: Is this available in current SVN ? Changes are always checked into SVN first and then made available in a tag. Leif. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] ChannelRedirect
Hello, I am implementing a small ACD system on Asterisk 1.6.2.17.2 I need help with ChannelRedirect. I have a caller connected to an agent. The agent may request additional help by consulting another department. I can't use manual process with blind or directed transfer as the agent have many different numbers to dial. The message with the proper dial number is coming from the host. I got that handled in my application as well. but while I know the channels for agent and caller, I can't seem to get ChannelRedirect to work properly for me. I am using Dual ChannelRedirect with AMI interface by taking the caller port and directing the call to a predefined conference bridge. The other channel needs to be redirected to an outside number. For some reason, I have both channels going to the same number. I am not sure if I am specifying the right channels in ChannelRedirect. I am not married to AMI approach either. I can use AMI to Redirect channels to a dialing plan and handle everything in the dialing plan as well. It just seemed it was easy to use the dual ChannelRedirect. Please let me know what is the best way to handle this condition. I will also need to have an ability to conference the caller, agent and outside party if the agent requests that. It would be a great help to get the steps for that as well. thanks in advance. If I miss any crucial information, please let me know and I will post that Alex -- _ -- 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] Free CNAM
On 5/29/2011 8:55 AM, Richard Kenner wrote: What happens when the CNAM is changed? How often does it go back and poll the database? That's actually a very very good question! Are entries in the database given a TTL/Expiration before being checked/researched again? Slainte, Sherwood McGowan -- _ -- 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] Free CNAM
Cool topic! Our company (generationD) developed some CID scripts for free use, and we would be interested in building and hosting this service. On the spec side, how do we avoid users claiming numbers belonging to others? (Could be an admin nightmare) Do we allow number ranges? Do we require caching, or limit lookups? (If not then this can get real expensive to host real fast) Just some ideas... From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Skyler [skchopper...@gmail.com] Sent: Thursday, June 02, 2011 3:38 PM To: Asterisk Users List Subject: Re: [asterisk-users] Free CNAM Hi all, Let’s get some feedback going here and see if there is any general support in a user-driven CNAM concept. Assuming that your landline/mobile outbound provider does not push caller-name + number for you with your calling plan. Would you pay $1/yr to have the access to update your own personal CNAM info in a database that you can trust to be correct? One that 1000’s or even 100,000’s of other voip/pbx owners will use? S. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Thursday, June 02, 2011 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Free CNAM I just checked several of my numbers and several others known to me, it really isn't much better 2 of them returned names other than mine, and all had the wrong city, though at least the state was correct. All but one also had the wrong carrier. I fear these databases are are so full of errors that they are mostly worthless. John Novack Pascal Bruno wrote: If you can use curl, and can do some text parsing and know regular expressions, you may be able to use this free CNAM service: http://www.numberguru.com/ and integrate into your system. This one appears to have a more complete database. When I tried my number, I have gotten my full name, but when I use the FreeCNAM project below, I just get Florida. On Wed, Jun 1, 2011 at 8:11 AM, Michael R. Wally michael.r.wa...@gmail.commailto:michael.r.wa...@gmail.com wrote: I've been toying around with the idea of starting some kind of 'Open CNAM' project to destroy the current money hustle BS that dominates this industry. The ever-growing FreeCNAM database may be a good starting point for such a project. I would also like to use Bitcoin (BTC) as the micropayment solution for user-requested updates. Some nominal fee. If anyone wants to get involved, contact me. On 06/01/2011 07:51 AM, Skyler wrote: Hi, The junk in CNAM databases like FLORIDA, ONTARIO etc. is IMO the carrier's way to isolate their users and another excuse to charge more money for 'the better plan'. In the end, it's the carrier that inputs the info so if it shows FLORIDA with one database I can't see how any other database would be different as the carrier is the only one that controls the outbound CID info. Calling me from POTS to snatch the CID will result in the same. ...unless there were a user friendly CNAM service, where info could be updated by the end-user and queried freely by voip providers. I would update my cellular numbers for sure and know at least a dozen people that would do the same. Everyone is going VoIP so why not? Talking about 'where's the money or angle'... here is one, vanity. Charge $1/yr to a user per DID, if I don't renew then delete it and re-query the original carrier. 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 -- Pascal B. Personal Web Sitehttp://www.pascalbruno.com/ Twitter: @petchaw -- _ -- 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 -- Dog is my Co-pilot No virus found in this message. Checked by AVG - www.avg.comhttp://www.avg.com Version: 10.0.1375 / Virus Database: 1511/3675 - Release Date: 06/02/11 -- _ -- 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:
[asterisk-users] chan_dahdi.c, dtmfmute, rtp.c
Hello, I am searching for a DTMF issue on my setup ( 2 years and counting ), and I am wondering why rtp.c has code to mute DTMF ( the rtp-dtmfmute variable ), but this same mechanism does not exist in dahdi. I am sending a DTMF over SIP w/ RTP RFC2833 to the asterisk box with the dahdi card. The dahdi card sends it out on the PRI line. Trouble is, the DTMF is echoed back and the echo canceller doesn't catch it. So asterisk thinks there is a an inbound DTMF happening : [2011-06-02 21:05:41.847] DEBUG[1333] rtp.c: - RTP 2833 Event: 0004 (len = 4) [2011-06-02 21:05:41.847] DEBUG[1333] rtp.c: Sending dtmf: 52 (4), at 192.168.1.5 [2011-06-02 21:05:41.847] DTMF[1333] channel.c: DTMF begin '4' received on SIP/siptest-000c [2011-06-02 21:05:41.847] DTMF[1333] channel.c: DTMF begin passthrough '4' on SIP/siptest-000c [2011-06-02 21:05:41.847] DEBUG[1333] chan_dahdi.c: Started VLDTMF digit '4' [2011-06-02 21:05:41.847] DEBUG[1333] rtp.c: - RTP 2833 Event: 0004 (len = 4) [2011-06-02 21:05:41.960] DEBUG[1333] chan_dahdi.c: Exception on 66, channel 50 [2011-06-02 21:05:41.960] DEBUG[1333] chan_dahdi.c: Got event Event 131124(131124) on channel 50 (index 0) [2011-06-02 21:05:41.960] DEBUG[1333] chan_dahdi.c: DTMF Down '4' [2011-06-02 21:05:41.960] DTMF[1333] channel.c: DTMF begin '4' received on DAHDI/50-1 [2011-06-02 21:05:41.960] DTMF[1333] channel.c: DTMF begin passthrough '4' on DAHDI/50-1 I looked through rtp.c and see code that handles this in the case of rtp to rtp, it says Ignore potential DTMF echo I also looked at chan_dahdi.c the only way to ignore DTMF is with disable_dtmf_detect() which seems to be used when Native bridging, so it's not what I am looking for. I found the chan-dtmf_tv variable in channel.c which specifies a gap between DTMF, but it requires 45 milliseconds, in theis case the echo is 113 milliseconds later, and on a different channel, so that doesn't drop the duplicate DTMF. How can I tell asterisk that the DTMF coming back is an echo and not a new DTMF ? The problem I am having is that later on, the echoed DTMF is causing real DTMF's from the user to be Ignored potential DTMF echo It's got it backwards. It's deteching the echo as a real DTMF and the real dtmf as echo. I tried both mg2 and oslec echo canceller, I saw no difference between the two. What is the next step in debugging this issue ? David -- _ -- 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] benefits of asterisk 1.8
Paul, With due respect to Digium work, are there no issues with Asterisk 1.8? https://issues.asterisk.org/view_all_bug_page.php [SATISH] On Thu, Jun 2, 2011 at 9:21 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 06/02/2011 10:29 AM, Eric Wieling wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Thursday, June 02, 2011 11:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] benefits of asterisk 1.8 On 11-06-02 09:35 AM, vip killa wrote: what do you mean Asterisk 1.8 is not stable enough yet? Can you give specific examples/scenarios? I too would like to see a specific example, additionally if you can create an test using the testsuite I'll be happy to review it and merge the code into subversion. -- _ -- 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] How to continue processing a context after a Hangup
Use Asterisk Application 'System()' in h extension to create callfile which will handle your callback. You can also try for 'Originate()' application. [SATISH] 2011/6/3 Antonio Modesto mode...@isimples.com.br Good afternoon, I'm trying to write a simple callback context, but i need to hangup an incoming call and then call the origin number back, the problem is that asterisk stops processing the call after Hangup() application then it is not able to dial the origin number back. Sorry for the grammatical erros. -- _ -- 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] How to continue processing a context after a Hangup
Warren,
A good example given.
Just suggest to use 'Move' instead of 'Copy' for placing callfile in
outgoing folder.
A J Stiles has explained it in a better way in one of his replies.
http://lists.digium.com/pipermail/asterisk-users/2011-May/262929.html
[SATISH]
On Fri, Jun 3, 2011 at 1:16 AM, Warren Selby wcse...@selbytech.com wrote:
2011/6/2 Antonio Modesto mode...@isimples.com.br
Good afternoon,
I'm trying to write a simple callback context, but i need to hangup an
incoming call and then call the origin number back, the problem is that
asterisk stops processing the call after Hangup() application then it is not
able to dial the origin number back.
The way I did it was to use a DeadAGI from the 'h' exten that created a
call file. This is how I did it for a client on Asterisk 1.4.x:
[rec-call-back-in]
exten = new,1,Answer()
exten = new,n,Wait(1)
exten = new,n,Playback(vm-intro)
exten = new,n,Playback(beep)
exten = new,n,Set(timestamp=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten = new,n,Set(FILENAME=reccallback/${CALLERID(num)}-${timestamp})
exten = new,n,Record(${FILENAME}.gsm,,,q)
exten = new,n,Playback(vm-goodbye)
exten = new,n,Hangup()
exten = h,1,Verbose(Hangup after recording)
exten = h,n,DeadAGI(reccallback.agi,${FILENAME},${TIMESTAMP})
[rec-call-back-out]
exten = out,1,Wait(2)
exten = out,n,Playback(${playbackfile})
exten = out,n,Hangup()
reccallback.agi:
#!/usr/bin/perl
use Asterisk::AGI;
use File::Copy;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
my $callerid = $input{'callerid'};
my $recfile = $ARGV[0];
my $timestamp = $ARGV[1];
open CALLFILE, /var/spool/asterisk/tmp/$callerid-$timestamp.call;
if (length($callerid) 4) {
print CALLFILE Channel: SIP/external-sip-provider/+1$callerid\n;
} else {
print CALLFILE Channel: SIP/$callerid\n;
}
print CALLFILE CallerID: \CUSTOMER\ XX\n;
print CALLFILE MaxRetries: 2\n;
print CALLFILE RetryTime: 60\n;
print CALLFILE WaitTime: 20\n;
print CALLFILE Context: rec-call-back-out\n;
print CALLFILE Extension: out\n;
print CALLFILE Priority: 1\n;
print CALLFILE Set: playbackfile=$recfile\n;
close CALLFILE;
sleep(5);
copy(/var/spool/asterisk/tmp/$callerid-$timestamp.call,
/var/spool/asterisk/outgoing/$callerid-$timestamp.call) or die copy
failed: $!;
exit;
--
Thanks,
--Warren Selby, dCAP
Our website just got a facelift! Check it out!
http://www.SelbyTech.com 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:
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
asterisk-users mailing list
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