Re: [asterisk-users] What is wrong in m
The problem is the OP never performs a Fetch of the data returned by the
Query...
From the VoIP-info page for Cmd MYSQL
MYSQL(Query resultid ${connid} query-string)
Executes standard MySQL query contained in query-string using established
connection identified by ${connid}. Result of query is stored in ${resultid}.
MYSQL(Fetch fetchid ${resultid} var1\ var2\ ...\ varN)
If any rows are available to select, ${fetchid} is set to 1 and a single row is
fetched from a result set contained in ${resultid}. The return fields are
assigned to ${var1}, ${var2} ... ${varN} respectively. If no rows are left to
select, ${fetchid} is set to 0 and ${var1}, ${var2} ... ${varN} remain
unchanged. --
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Re: [asterisk-users] Asterisk Online Training
+1...I am an autodidact myself, never took any courses in IT or Telephony other a computing course in tge late 80s that was actually a typing class that used computers. Slainte, Sherwood McGowan Sent from my iPhone On Jun 6, 2011, at 5:58 PM, Amadu alsta...@gmail.com wrote: Way to go Steve. That's the best way to learn. Steve Totaro stot...@asteriskhelpdesk.com wrote: 2011/6/6 Antonio Modesto mode...@isimples.com.br: Good Morning, I'm thinking about buying the asterisk six-months online course, Have somebody here that bought that course? What is your opinion? Thanks. I have not bought the course nor will I. I am self taught everything in IT and Telephony. I know each person learns differently, that is why when I train someone, I don't show them, I make them do it and coach until they don't need help. In college, I studied for business administration and accounting. It was too easy. My classes at WVU had 300 students, so attendance was impossible to keep track of. I aced all the tests. Then one summer I was a book keeper at t a travel agency. I hated it. I hated waking up to go to work, I hated the commute, and I hated the job. Then it dawned on me that I was always good at computers and figuring things out, ever since I got my VIC 20 and later my Commodore 64. I even had a Timex Sinclare(sp?) I got into a bit of trouble but since I was a kid and there was no malicious intent, the legal anchorites went away, just had to deal with my parents. Total tangent, I apologize. Classes slow me down and then become super boring. I fell asleep in the 3Com NBX course a few times, but got a 98% on the cert exam. I learned Cisco on the job, walked into a Prometric and passed with 80 something % and got my CCNA. Same for M$ products. I am planning to take the Red Hat cert soon. Anyways, I find structured courses to be a waste. Usually they hamper your creativity and are filled with fluff. If you cannot get a solid grasp of Asterisk in six months on your own lab using Google, voip-info, and howtos, then the course is probably for you you. Thanks, Steve Totaro -- _ -- 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] Pops clicks at the end of sound files
On Jun 6, 2011, at 10:51 PM, Steve Edwards asterisk@sedwards.com wrote: Sox has a bunch of obtuse (IMNSHO) commands. There may be one that could automagically trim the pop for you. The argument is question is the trim command. If the OP wishes to find an automagic method, they would need to determine the length of the file in seconds, then feed the length - durationofthepop to the sox trim command -- _ -- 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 Calculation
May I add...I still have documented cases of asterisk 1.4.x running ulaw with no transcoding and running 2k+ concurrent calls on a CentOS 4(5?, fuzzy) machine with 2ghz CPU and 2gb ram Sent from my iPhone On Jun 5, 2011, at 3:02 PM, Steve Edwards asterisk@sedwards.com wrote: On Sun, 5 Jun 2011, Khaled W. Chehab wrote: 1-Is there a calculator I can download for that 2-What I the maximum simultaneous calls that can asterisk handle using CPU 3.0 MHZ and 4GB ram With rtp g729 and there is no codec transcoding 3-And what is the number of simultaneous calls if I use direct RTP (Canreinvite=no /Directrt=yes) 1) No. Because every case is a bit different and nobody has taken the time to research and document it. 2) In the 'hundreds.' I have a 5 yr old 3.4 Xeon server with 2GB of ram running all kinds of AGIs that handles 300 simultaneous ULAW calls without issue and without any 'tuning.' The Asterisk process uses less than 100MB so more GBs means nothing. 3) Probably in the thousands depending on what those calls are doing. (Just guessing here because I have no experience with this configuration.) Would a SIP server like OpenSIPS be a better platform choice? More details will yield better responses. -- 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 -- _ -- 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 6/3/2011 9:49 AM, satish patel wrote: But unfortunately i compiled with DON'T OPTIMIZED option do you think it will generate dumpcore in that case ? Yes, it will create a coredump. Telling the compiler to not optimize (IIRC) leaves more debugging info in the binary for dumps -- _ -- 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
I don't know the statistics involved, but not allowing the compiler to optimize would almost assuredly have some negative effect on performance Sent from my iPhone On Jun 3, 2011, at 10:16 AM, satish patel satish...@hotmail.com wrote: But anyway let me set coredump=yes in asterisk.conf Do you think its a good idea to compile with DON'T OPTIMIZED option in production ? does it impact on performance ? -S CC: asterisk-users@lists.digium.com From: sherwood.mcgo...@gmail.com Date: Fri, 3 Jun 2011 10:13:31 -0500 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] benefits of asterisk 1.8 No, it just means that the coredump will not have information that is as useful Sent from my iPhone On Jun 3, 2011, at 10:02 AM, satish patel satish...@hotmail.com wrote: Sherwood, I was wrong here But unfortunately i compiled with DON'T OPTIMIZED option do you think it will generate dumpcore in that case ? I have just cross check and we have option OPTIMIZED. That mean don't create coredump right ? -S Date: Fri, 3 Jun 2011 09:53:01 -0500 From: sherwood.mcgo...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] benefits of asterisk 1.8 On 6/3/2011 9:49 AM, satish patel wrote: But unfortunately i compiled with DON'T OPTIMIZED option do you think it will generate dumpcore in that case ? Yes, it will create a coredump. Telling the compiler to not optimize (IIRC) leaves more debugging info in the binary for dumps -- _ -- 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 -- _ -- 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] asterisk fails when DNS or internet fails
True, but with all due respect, if the cache's TTL expires and the OP's PBX cannot reach an external DNS server, they have bigger problems ;-) Slainte all! The Mick Sent from my iPhone On May 30, 2011, at 9:41 PM, Mark Deneen mden...@gmail.com wrote: On Mon, May 30, 2011 at 2:44 AM, gincantalupo gincantal...@fgasoftware.com wrote: Hi, it is a known problem, one of the worst. To avoid it: - do not use urls, only ip addresses in sip.conf or put your urls inside /etc/hosts (is what I do especially sip providers urls) or install a dns-cache on your pbx (maybe the best solution) Giorgio Even a dns cache won't help you forever. Once the entry's TTL expires, your cache won't be able to give you an answer until it can contact root servers on the internet. -M -- _ -- 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 1..8 multiple queue
It's REALLY not that hard to emulate the old AgentCallbackLogin, or to use AddQueueMember, I'm still trying to understand the OP's issue. Is this just because you have to write a little new dialplan code, the fact that it doesn't work the same as before (a common complaint when things change of course), or something else that I am just not seeing? Sent from my iPhone On May 26, 2011, at 4:50 AM, Lenz Emilitri lenz.lo...@gmail.com wrote: Shameless plug: the QueueMetrics agent page, even in the free 2-agent version, can emulate this behavior. You may want to check it out. l. 2011/5/25 satish patel satish...@hotmail.com Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -S -- Loway - home of QueueMetrics - http://queuemetrics.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] Is this Asterisk issue of feature
Wait is not ONLY for use with Answer, that just happens to be a common use.
To answer the OP, a quick look at
http://www.voip-info.org/wiki/view/Asterisk+cmd+Wait confirmed that Wait does
not work after a channel is hung up:
If Wait() is run on a channel which has been hung up (i.e. from the 'h'
extension), the Wait() application returns 0 immediately and no further
processing of the 'h' extension priorities takes place. You can get around this
with 'System(path/to/sleep Xs)', where 'X' is the number of seconds to wait.
Nasty, but it works.
Sent from my iPhone
On May 26, 2011, at 6:00 AM, James zhu zhulizh...@live.com wrote:
hi:
i think the wait is used for answer command. please show any debug info?
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards,
gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
Date: Thu, 26 May 2011 16:15:31 +0530
From: virbh...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Is this Asterisk issue of feature
Hi List,
I am confuse about this feature.
When we use Wait(20) in active call session then it's work. But when we use
after hangup the call then Asterisk don't wait from define time.
Ex:-
[call_log]
exten = 4368,1,Answer()
exten = 4368,n,Flite(Welcome)
exten = 4368,n,Set(__StartTime=${STRFTIME(${EPOCH},Asia/Calcutta,%Y-%m-%d
%H:%M:%S)})
exten = 4368,n,Set(__uniqueId=${UNIQUEID})
exten = 4368,n,Wait(20) ; At this moment it's work.
exten = 4368,n,Hangup()
exten = h,1,NoOp(***Now wait(20) wouldn't work
* )
exten = h,n,Wait(20) ; At this moment it's not work.
exten = h,n,NoOp(***never come into execution
**)
-
Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
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Re: [asterisk-users] Is this Asterisk issue of feature
sleep is a Bash command, the author was illustrating that you must include
the full path to the binary
Sent from my iPhone
On May 26, 2011, at 8:07 AM, virendra bhati virbh...@gmail.com wrote:
Hi ,
Thanks for reply ..
What is the meaning of that line which you have mention on the recent
conversation
System(path/to/sleep Xs)
path/to/sleep = is the paths of any php script where sleep function is
mention or anything else ?
please illustrate if it is possible
On Thu, May 26, 2011 at 4:43 PM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
Wait is not ONLY for use with Answer, that just happens to be a common use.
To answer the OP, a quick look at
http://www.voip-info.org/wiki/view/Asterisk+cmd+Wait confirmed that Wait does
not work after a channel is hung up:
If Wait() is run on a channel which has been hung up (i.e. from the 'h'
extension), the Wait() application returns 0 immediately and no further
processing of the 'h' extension priorities takes place. You can get around
this with 'System(path/to/sleep Xs)', where 'X' is the number of seconds to
wait. Nasty, but it works.
Sent from my iPhone
On May 26, 2011, at 6:00 AM, James zhu zhulizh...@live.com wrote:
hi:
i think the wait is used for answer command. please show any debug info?
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards,
gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
Date: Thu, 26 May 2011 16:15:31 +0530
From: virbh...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Is this Asterisk issue of feature
Hi List,
I am confuse about this feature.
When we use Wait(20) in active call session then it's work. But when we use
after hangup the call then Asterisk don't wait from define time.
Ex:-
[call_log]
exten = 4368,1,Answer()
exten = 4368,n,Flite(Welcome)
exten = 4368,n,Set(__StartTime=${STRFTIME(${EPOCH},Asia/Calcutta,%Y-%m-%d
%H:%M:%S)})
exten = 4368,n,Set(__uniqueId=${UNIQUEID})
exten = 4368,n,Wait(20) ; At this moment it's work.
exten = 4368,n,Hangup()
exten = h,1,NoOp(***Now wait(20) wouldn't work
* )
exten = h,n,Wait(20) ; At this moment it's not work.
exten = h,n,NoOp(***never come into execution
**)
-
Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
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update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
--
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Re: [asterisk-users] Tr : how to user SIP realtime option
I have configured tons of 1.4.x installs with various levels of realtime integration. Are you asking about the specific sub-version 1.4.26.1, or the 1.4 branch in general? Also, in order for anyone to help you, we'd need some info like how ou set up realtime, log outputs, etc Sent from my iPhone On May 25, 2011, at 4:33 PM, Dr Ox dr_o...@yahoo.fr wrote: Hi Guys, Has anyone been able to configure the realtime using Asterisk 1.4.26.1? So far I've successfully configured the realtime using Asterisk 1.6.2 but while trying to use 1.4, Asterisk keep restarting. Any clue or suggestion? -- _ -- 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] Tr : how to user SIP realtime option
On 5/25/2011 4:33 PM, Dr Ox wrote: Hi Guys, Has anyone been able to configure the realtime using Asterisk 1.4.26.1? So far I've successfully configured the realtime using Asterisk 1.6.2 but while trying to use 1.4, Asterisk keep restarting. Any clue or suggestion? By the way, I forgot to add, I'd be more than happy to help you with your SIP realtime integration. Please post the relevant configuration files (such as extconfig.conf, odbc.ini odbc-inst.ini, etc) and a mysqldump of structure and data from your sip table(s). Full log output from your server would also be helpful. You may reply to me offlist if you like -- _ -- 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 multiple queue
On 5/25/2011 12:32 PM, satish patel wrote: Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -S Use of the realtime architecture for queue members is my preferred method. -- _ -- 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 multiple queue
Yes, there are other ways, I was only offering the solution that has worked best for me. Keep in mind, you are not limited to MySQL for realtime, Asterisk can use any ODBC DSN for the data backend. Oracle, Access, MSSQL are all examples, if I recall correctly you can even connect SQLite and DB2. However, let me ask you this...what trouble are you having with AddQueueMember and it's related applications that is making it hard for you? Sent from my iPhone On May 25, 2011, at 7:20 PM, Satish Patel satish...@hotmail.com wrote: Thanks for reply but is there any alternative way? Because we don't have mysql and we dont want to use mysql. -- Sent from my iPhone On May 25, 2011, at 6:43 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On 5/25/2011 12:32 PM, satish patel wrote: Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -S Use of the realtime architecture for queue members is my preferred method. -- _ -- 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] When someone helps you, at least let them know if the problem is resolved or not
Because poking the sleep tiger is fun for some, especially if you're just BARELY faster than the tiger ;-) *poke* On Tue, May 17, 2011 at 5:26 AM, Andrew Thomas a...@datavox.co.uk wrote: And why would you post a reply 5 days after my last post - and 4 days after the threads last one? Do you want to keep this thread going? I suggest letting it die on it's own. _ -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: 17 May 2011 02:05 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not Seriously guys. Why would anyone other than the two of you need to read this. It's a personal conversation. We all know who you both are and your achievements etc. The longer the conversation goes on the more off topic it becomes :-) -- 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 If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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 -- Sherwood McGowan Telecommunications and VOIP 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] When someone helps you, at least let them know if the problem is resolved or not
Definitely, especially with a nice chianti and some fava beans...*slurping sound* On Wed, May 18, 2011 at 6:19 PM, Matt Riddell li...@venturevoip.com wrote: On 17/05/11 5:24 PM, Sherwood McGowan wrote: I like puppies Yeah, much more tasty than fully grown dogs :-P -- 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 -- Sherwood McGowan Telecommunications and VOIP 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] When someone helps you, at least let them know if the problem is resolved or not
I like puppies On Mon, May 16, 2011 at 8:05 PM, Matt Riddell li...@venturevoip.com wrote: Seriously guys. Why would anyone other than the two of you need to read this. It's a personal conversation. We all know who you both are and your achievements etc. The longer the conversation goes on the more off topic it becomes :-) -- 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 -- Sherwood McGowan Telecommunications and VOIP 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] When someone helps you, at least let them know if the problem is resolved or not
Wow...somehow this turned into a something so much darker than the original intent*sits back and watches the show* Thanks guys, that little mini bonfire made an otherwise boring day into an entertaining Asterisk-Users version of WWE Raw. Cheers! 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] Tricky: Progress, Delay, DTMF / background calling
Try reading up on Local channels, it will accomplish everything you wish.
On Wed, May 11, 2011 at 8:59 AM, Markus unive...@truemetal.org wrote:
Hi again,
no one got an idea? :-( Or did my request not make any sense? Or is the
answer to obvious that no one bothers to reply? :-)
Thanks again!
On a second thought, I don't need the predetermined delay. I can probably
just set that with additional w's in the DialBackground command (which I
made up).
So rather something like:
_X.,1,Progress
_X.,2,DialBackground(SIP/123456@provider
,,D(ww${EwwXwwTwwEwwN}ww))
_X.,3,ConnectLegs
Thanks again.
Hi,
has the following been done before respectively is it possible with
Asterisk? I searched the archives but couldn't locate anything.
1. Call to comes in via SIP.
2. Call is not answered yet but progress continues.
3. At the moment the call comes in something like this gets spawned in
the
background:
Dial(SIP/123456@provider,,D(ww${EXTEN})
which should translate to:
Dial(SIP/123456@provider,,D(ww)
But even better would be take the ${EXTEN} and put some w's between
them:
Dial(SIP/123456@provider,,D(ww5ww5ww5ww5)
4. After a pretermined amount of time since the call came in
respectively
the Dial command was spawned in the background, e.g. 15 seconds,
Asterisk answers the call and the call legs are connected together.
So, with some fantasy commands, something like this:
_X.,1,Progress
_X.,2,DialBackground(SIP/123456@provider
,,D(ww${EwwXwwTwwEwwN}),ANSWER-AND-CONNECT-LEGS(15)
I hope my request is not too cryptic. In short: I'd like to receive
calls
to arbitrary extensions, but not answer them directly, only after a Dial
command has been spawned and a predetermined amount of time has passed
since the Dial command has been spawned / since the Dial command has
connected to 123456.
Possible?
I'm new to the list, hi! :)
Thank you!
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Re: [asterisk-users] 40sec between dial execution and sending SIP request
Good call Warren, might I add that a great idea would be to set debug and verbose to 5, change the timestamp format on your logs temporarily to show HH:mm:ss:ms (don't necessarily need milliseconds, but I'm an accuracy geek), make sure you have a log that is writing ALL output (except maybe DTMF, but error, warning, info, debug, verbose are all necessary) then do a logger reload and a logger rotate, dial your test call, and then attach the resulting logfile. On Tue, May 10, 2011 at 2:28 AM, Warren Selby wcse...@selbytech.com wrote: Show us the cli trace of the delay. Thanks, --Warren Selby, dCAP On May 10, 2011, at 2:18 AM, Pezhman Lali l...@lopl.net wrote: thanks, this delay is occurred on asterisk server, between dial execution and CALLED . On Mon, May 9, 2011 at 7:12 PM, Warren Selby wcse...@selbytech.com wcse...@selbytech.com wrote: On Mon, May 9, 2011 at 7:26 AM, Pezhman Lali l...@lopl.net l...@lopl.net wrote: Dear I have a small pbx with asterisk 1.6.2.16. I have a funny problem, there is exactly 40sec between dial execution and sending first invite packet on sip. do you have any idea where the problem is ? Check the dial timeout on your phone itself. What model phone do you have? -- Thanks, --Warren Selby, dCAP http://www.selbytech.comhttp://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Pezhman Lali -- _ -- 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 http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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 -- Sherwood McGowan Telecommunications and VOIP 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] When someone helps you, at least let them know if the problem is resolved or not
I'll keep this brief because I don't want to come across like any more of an a$$ than I absolutely have to, especially since I know I've blown my stack before. Gentlemen (and Ladies, if you're out there), If someone gives you advice on this list, and ESPECIALLY if they give you advice offlist, have the courtesy to (AT THE LEAST) to let them know when/if you get your question answered or your problem solved. As many people point out, on community supported mailing lists and forums around the world, these user lists are comprised of people who are giving their time freely to help others learn about the software the list is about. Sometimes those lists are about software that is quite useful in a commercial setting, perhaps even very much in demand, like Asterisk. Now, you should always appreciate when you get assistance from people on user lists, but when you're asking for help on a list like this one, (where I'd say 80% of the participants on the list are professionals who earn their living by selling their knowledge of how to install, configure, and maintain a server application like Asterisk) it would be extremely appreciated if you show some courtesy to the individual(s) who assisted you for free. I've had several individuals contact me offlist (without being given permission first, which is first and foremost bad form) and ask for my assistance with configuring a feature, troubleshooting an issue, and once I got an email that said something along the lines of: I saw a post on the list where you said you could accomplish * insertNiftyFeatureThatDidNotPreviouslyExistHere* Tell me how to do it I'm sure many of you have been the recipient of more than your fair share of emails offlist asking for help, and I'm sure a great number of you try to offer assistance. What is bothering me is the fact there seems to be a new trend forming, wherein I don't get a repsonse from the person I tried to help, even when I can feel confident in saying that I know I gave them the piece of information they needed in order to answer their question and accomplish the goal of making Asterisk perform the way they wanted. Has anyone else noticed this trend? Those of you who are making the requests, is there a reason why you don't feel the need to be courteous and at least say, Hey that advice worked, everything's working now? Next time you ask for help, especially when it's offlist (and even MORE SO when you're contacting someone you weren't invited to contact offlist), I want you to remember that the person you're contacting usually gets paid for their time as an Asterisk professional, and that they're helping you for free. Hell, if you want to get down to brass tacks about it, thatr person who is taking the time to try and help you is increasing his or her own professional competition.. that's all...nothing super rude, but I had to get that one out there I get annoyed when I answer about 12-13 questions (all in separate emails, mind you) from someone, and then I never get even find out if I was successful in helping them -- Sherwood McGowan Telecommunications and VOIP 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] When someone helps you, at least let them know if the problem is resolved or not
On Tue, May 10, 2011 at 5:38 PM, Nic Colledge n...@njcolledge.net wrote: +1 I agree with all of this. Just wanted to add that there’s another good reason for not contacting people off-list and that is when you do so you leave no record of your conversation that others can use in future to solve the same problem. You can learn a lot just by reading the archives. Nic. Very good point Nic! Oh and thanks for the +1...Usually when I get on a rant, I piss everyone off ;-) -- Sherwood McGowan Telecommunications and VOIP 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] When someone helps you, at least let them know if the problem is resolved or not
+1 from me too. The other thing is that when you answer to say the problem has been solved this goes into the archives meaning that people can use Google to answer their own questions rather than having to even ask the list. There have been times when I've searched for a solution to a problem, found like 10 answers, and nobody has said whether they work or not so you have to try all of them. -- Cheers, Matt Riddell Believe me mate, I feel you, on that note. Not only because of my time when I was asking more questions than I was answering, but also from the standpoint of wishing the answers were a little more prevalent for the searching party to find so that I didn't see s many repeats on the list ;-) Cheers guys! -- Sherwood McGowan Telecommunications and VOIP 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] SIP secruity: username and password
Little to none...SIP is set up so that the packet contains identifiable data (the username) but the authentication is performed with a digest of the username password [domain] and [CalliD] (I think I got that right) On Thu, May 5, 2011 at 7:08 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; When the endpoint register on Asterisk or initiate a call, so they exchange the sip username and password. What is the possibility that this will be capture by the hacker and how to avoid this problem? 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 -- Sherwood McGowan Telecommunications and VOIP 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] SIP secruity: username and password
Thanks Alex for clearing up the bit about the NONCE, that's what I was trying to remember when I said CallID :) Good explanation by the way! :) On Thu, May 5, 2011 at 7:17 AM, Alex Balashov abalas...@evaristesys.comwrote: Bilal, On 05/05/2011 08:08 AM, bilal ghayyad wrote: When the endpoint register on Asterisk or initiate a call, so they exchange the sip username and password. What is the possibility that this will be capture by the hacker and how to avoid this problem? Strictly speaking, there is no inherent connection between either registration or call initiation on the one hand, and authentication. Both of those scenarios can be performed in an authentication-free fashion. In fact, in most cases the SIP UAC will first attempt to send both a REGISTER and an INVITE request without any authentication credentials. However, it is typical of a SIP UAS providing retail services to the public at large to reply to those requests with a 401 or 407 proxy challenge requesting authentication. The UAC then resends the request with digest authentication headers, including a password encrypted via a cryptographic one-way hash function. The entire mechanism was borrowed from HTTP digest authentication. The authorisation username can absolutely be intercepted, as it is transmitted it in plain text. But this is not news. The password is encrypted, and while the encrypted version can be intercepted, it is encrypted using a one-time nonce value that is part of the 401 or 407 challenge sent by the UAS. Nonce values typically have fairly stringent expiration times, at least on good implementations, but nonce replay attacks are possible in principle. This mechanism is reasonably secure, as a compromise with the interoperability requirements of providing SIP service across the public Internet. In high-stakes situations, however, it may not be sufficient, and may call for SIP over a TLS transport, or encrypted tunnels. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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 -- Sherwood McGowan Telecommunications and VOIP 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] missed call notification
? 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 -- Sherwood McGowan Telecommunications and VOIP 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] missed call notification
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 -- Sherwood McGowan Telecommunications and VOIP 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 -- _ -- 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 -- Sherwood McGowan Telecommunications and VOIP 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] missed call notification
Heheh, well Warren, I'm just a quick draw I guess ;-) Hey, at least you have
dCAP by your name! I've been at this 6-7 years and still haven't gotten off
my butt and taken the tests :D
On Thu, May 5, 2011 at 1:20 PM, Warren Selby wcse...@selbytech.com wrote:
And Sherwood beats me to the punch again :).
Thanks,
--Warren Selby, dCAP
On May 5, 2011, at 1:15 PM, Sherwood McGowan sherwood.mcgo...@gmail.com
wrote:
No, the variables are channel specific except for when they're inherited,
which doesn't affect you here
On Thu, May 5, 2011 at 1:02 PM, satish patel satish...@hotmail.com
satish...@hotmail.com wrote:
After google i found something and i tried following. I set variable
before Dial and its give me proper value in h extension but now question
is if multiple user dial multiple extension then will it overwrite current
variable value ?
exten = s,1,Set(_CALLED_EXT=${ARG2})
exten = s,n,Dial(${ARG2}iax2/${ARG1},20,t)
--
From: satish...@hotmail.comsatish...@hotmail.com
To: asterisk-users@lists.digium.comasterisk-users@lists.digium.com
Date: Thu, 5 May 2011 17:52:54 +
Subject: Re: [asterisk-users] missed call notification
Could you please tell me how ( Syntax ) and where in macro ?
I am not expert in dialplan variables. I appreciate your help
--
Date: Thu, 5 May 2011 12:44:19 -0500
From: sherwood.mcgo...@gmail.comsherwood.mcgo...@gmail.com
To: asterisk-users@lists.digium.comasterisk-users@lists.digium.com
Subject: Re: [asterisk-users] missed call notification
if you saved ${_CALLED_EXT} to the value of ${EXTEN} from within the
macro, you'd get 's'do it while you still have the called number as the
EXTEN
On Thu, May 5, 2011 at 12:42 PM, satish patel satish...@hotmail.com
satish...@hotmail.com wrote:
Also check for CANCEL, since this should be the status if the caller
hangs up before the call is picked up.
But CANCEL is return nothing
[macro-stdexten]
exten = s,1,Dial(${ARG2}iax2/${ARG1},20,t) ; Ring the
interface, 20 seconds maximum, call screening option (or use P for databased
call screening)
exten = s,n,Goto(s-${DIALSTATUS},1) ; Jump based on
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
;exten = s,n,Hangup()
exten = s-CANCEL,1,Verbose(Hangup call)
CLI
== Spawn extension (macro-stdexten, s, 1) exited non-zero on
'SIP/7527-0023' in macro 'stdexten'
== Spawn extension (from-sip, 7516, 1) exited non-zero on
'SIP/7527-0023'
Look like its going back to original extension :( I hate macro
--
From: satish...@hotmail.comsatish...@hotmail.com
To: asterisk-users@lists.digium.comasterisk-users@lists.digium.com
Date: Thu, 5 May 2011 17:15:53 +
Subject: Re: [asterisk-users] missed call notification
You want me to do this in macro-stdexten ? I have following dialplan. I
have used h extension in original context because you can't you h inside
macro right ?
[macro-stdexten]
exten = s,1,Dial(${ARG2}iax2/${ARG1},20,t) ; Ring the
interface, 20 seconds maximum, call screening option (or use P for databased
call screening)
exten = s,n,Goto(s-${DIALSTATUS},1) ; Jump based on
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s,n,Hangup()
exten = s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable,
send to voicemail w/ unavail announce
exten = s-NOANSWER,n,Hangup()
exten = s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send
to voicemail w/ busy announce
exten = s-BUSY,n,Hangup()
exten = s-CONGESTION,1,Voicemail(${ARG1},u) ; Like above,
write a macro for this case
exten = s-CONGESTION,n,Hangup()
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything
else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they press
*, send the user into VoicemailMain
[from-sip]
...blah...blah..
exten = h,1,System(/var/lib/asterisk/agi-bin/processcallemail.sh
${CALLERID(num)} ${CALLERID(name)} ${DIALSTATUS} ${VMSTATUS})
From: wcse...@selbytech.comwcse...@selbytech.com
Date: Thu, 5 May 2011 12:10:09 -0500
To: asterisk-users@lists.digium.comasterisk-users@lists.digium.com
Subject: Re: [asterisk-users] missed call notification
Set a variable ${_CALLED_EXT} to ${EXTEN} before you hang up the call,
then reference that variable in your h exten.
Thanks,
--Warren Selby, dCAP
On May 5, 2011, at 11:59 AM, satish patel satish...@hotmail.com
satish...@hotmail.com wrote:
Hi All,
I am using
http://www.theschmandts.org/blog/2007/05/05/email-notifications-for-missed-calls-in-asterisk/
http://www.theschmandts.org/blog/2007/05/05/email-notifications-for-missed-calls-in-asterisk/to
implement missed call feature. and i modify script to grab email address
from voicemail.conf
But i am not able to see DEST extension in this script
Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
On Thu, May 5, 2011 at 11:02 PM, Matt Riddell li...@venturevoip.com wrote: On 6/05/11 3:14 PM, Ira wrote: At 03:00 PM 5/5/2011, you wrote: Yes, but in my world there is one Atom powerd Linux box running Asterisk, 4 or 5 Windows machines and 2 Macs. If I want to test, it has to be on my production box and I'm more than happy to run beta software on that box. My comment is just that the protocol for me helping you is not clear to me. I have been beta testing since 1985 when I was able to crash Brief on the Novell network I used at work. Were you beta testing using your production servers then? Yes, I use my one and only server for testing. Brave and foolish soul that I am! :-) Fair enough then! -- 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 I Like Turtles. 'nuff said...get a beer! -- Sherwood McGowan Telecommunications and VOIP 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] Cordless VoIP Phones and Access Point hand-off?
ChanIsAvail + dialplan routing to call parking lot On Wed, May 4, 2011 at 6:02 PM, Ira i...@extrasensory.com wrote: At 03:21 PM 5/4/2011, you wrote: Barring that, if the cordless phone becomes un-reachable is there a way to automatically put the active call on hold, or park it? That's not the preferred solution, but it would work great until I figure something else out. Not that it applies but I recently installed a Snom M3 and it seems to behave like you want. When I walk out of range and then back in the call is usually still there. I've not tested past that so it might hang up after an unknown timeout. 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 -- Sherwood McGowan Telecommunications and VOIP 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] odbc error - server is gone
On Thu, Apr 28, 2011 at 11:09 AM, Rizwan Hisham rizwanhas...@gmail.comwrote: Hi list, yesterday I converted my voicemail.conf to realtime voicemail and also configured to store the voicemessages in a database using odbc as described here http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail and here http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage. I am using asterisk 1.4.2 with mysql. I also installed the proper odbc driver for mysql on the server. I successfully completed the conversion of a lot of voicemail users into db yesterday. But today on the CLI thsi error was showing; [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51 Driver][mysqld-5.0.68-log]MySQL server has gone away (70) [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51 Driver][mysqld-5.0.68-log]MySQL server has gone away (70) [Apr 28 11:40:54] WARNING[24676]: app_voicemail.c:2239 inboxcount: SQL Execute error! [SELECT COUNT(*) FROM voicemessages WHERE dir = '/var/spool/asterisk/voicemail/default/1757XXX/INBOX'] I know that the error is caused due to stale odbc connection with mysql. But i want to find out if there is a cure for it. Why the connection went stale in the first place also. Any ideas? -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.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 do you have sanitysql = select 1 configured in res_odbc.ini? -- Sherwood McGowan Telecommunications and VOIP 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] call files
On Sat, Apr 23, 2011 at 11:20 AM, Tiago Geada tiago.ge...@gmail.com wrote: Hi. Im having trouble setting variables in channel dialplan and re-using them in Extension dialplan... Im using the following call file: Channel: Local/210332450@ZonNew-Outbound CallerID: ZonNew-Outbound:49:210332450: MaxRetries: 5 RetryTime: 10 WaitTime: 60 Account: Outbound210332450 Context: agents Extension: 888210332450 Set: __PARTNER=ZonNew-Outbound Set: NUMBER=210332450 - In Local/210332450@ZonNew-Outbound I Set(bla='blabla'); It seems I cannot re-use this var in extension _888X in context agents... Basically the Channel dialplan has a Queue() and in _888X I would like to know the peer (or interface) that answered it... What can I do? Thanks in advance I'm a little confused by It Seems I cannot re-use this var in extension _888XX in context agentsOf course you can use it...but if you set bla to a different value in your code where your callfile is processed, Asterisk will (rightfully so) just set bla = to whatever you set it to Now, if the callfile doesn't send a channel through the context that you're trying to set blah, that's a little odd... Now, as far as retrieving the information about the interface that answered the calllook in queues.conf.samplethere's a nifty configuration option: *setinterfacevar=no ; (the default is no)* That option, when set to yes, causes several variables to be created *just*prior to the caller being bridged with the queue member... -- Sherwood McGowan Telecommunications and VOIP 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] missed call notification
On Fri, Apr 22, 2011 at 9:32 AM, satish patel satish...@hotmail.com wrote:
How this guys using h extension in macro
http://www.theschmandts.org/blog/?p=28
why its not working for me only. I am the caller and if i hangup then it
should parse h extension right ?
--
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Thu, 21 Apr 2011 18:16:00 +
Subject: Re: [asterisk-users] missed call notification
I am always googleing before putting anything here.. I was confused that's
why i came across to you guys! Still i am confused :(
-S
--
Date: Thu, 21 Apr 2011 13:01:52 -0500
From: sherwood.mcgo...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] missed call notification
On Thu, Apr 21, 2011 at 12:26 PM, satish patel satish...@hotmail.comwrote:
Hi,
I am looking at http://www.theschmandts.org/blog/?p=28 to setup missed
call notification but i am having issue. following is my dialplan
[macro-stdexten]
exten = s,1,Dial(${ARG2})
exten = s,2,Goto(s-${DIALSTATUS},1); Jump
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable,
send to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #,
return to start
exten = s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to
voicemail w/ busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they
press #, return to start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat
anything else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they
press *, send the user into VoicemailMain
exten = h,1,System(/var/lib/asterisk/agi-bin/processcallemail.sh ${ARG3}
${CALLERID(num)} ${CALLERID(name)} ${DIALSTATUS} ${VMSTATUS}
${EXTEN})
[from-sip]
exten = _7[0123]XX,1,macro(stdexten,${EXTEN},sip/${EXTEN})
Following CLI output look like its not executing h extension in
macro-stdexten. But if i add h extension in [from-sip] it works! do you know
why ?
-- Executing [7207@from-sip:1] Macro(SIP/7101-000a,
stdexten,7207,sip/7207) in new stack
-- Executing [s@macro-stdexten:1] Dial(SIP/7101-000a,
sip/7207) in new stack
== Using SIP RTP CoS mark 5
-- Called 7207
-- SIP/7207-000b is ringing
== Spawn extension (macro-stdexten, s, 1) exited non-zero on
'SIP/7101-000a' in macro 'stdexten'
== Spawn extension (from-sip, 7207, 1) exited non-zero on
'SIP/7101-000a'
-- Executing [h@from-sip:1] Hangup(SIP/7101-000a, ) in new
stack
== Spawn extension (from-sip, h, 1) exited non-zero on
'SIP/7101-000a'
All I can say is that you have scientific proof that the macro's h extension
is NOT being executed but the calling context's h extension *is*. Nevermind
why, that's more of a developer conversation. The reality is, you can
accomplish your task by putting the code you need in the calling context's h
extension.
--
Sherwood McGowan
Telecommunications and VOIP Consultant
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Meetme Time Limit?
On Thu, Apr 21, 2011 at 4:03 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.comwrote: Hi, You can use Meetme(1234,dL(1800)) where 1800 = 6 hours after 6 hours channel is hanf up regards Dhaval On Mon, Apr 18, 2011 at 9:31 PM, Bryant Zimmerman brya...@zktech.comwrote: Is there a way to place a hangup time on a dynamic Meetme conference. I am using Page() with a Meetme conf and I have had a few instances where someone from a wifi voip phone looses ip while doing a page and the page never hangs up. I have to kill it. I need to somehow limit the page so after a worse case 2Min timeout it hangs up. Thanks Bryant -- Dhaval's reply works for when you're running a MeetMe conference directly, which does not help Bryant (the question was phrased a little oddly, which caused the confusion I think) Regarding how to limit how long the Paging call can be, use the TIMEOUT(absolute) function. Here's an AEL example: [paging] exten = _92XX,1,Noop(Making sure the call only lasts 60 seconds or less) same = n,Set(TIMEOUT(absolute)=60); same = n,Page(insert page targets and options) Let me know if that works out for you! Regarding MeetMe time limiting in general, I'd like to add an alternative to Dhaval's solution, just to get it back out there in the intertubes so people can find it in the future. As of Asterisk 1.6 you can schedule RealTime MeetMe conferences. I've attached a structure dump of a table called conferences, just direct your extconfig.conf to use it for meetme, set schedule=yes in meetme.conf, and then set the start and end times in the table when creating a scheduled conference. Cheers all! Sherwood McGowan Coming soonSamuPBX scheduled_conferences.sql Description: Binary data -- _ -- 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] VoiceMail to text mail
On Wed, Apr 20, 2011 at 4:10 PM, Mark Deneen mden...@gmail.com wrote: On Wed, Apr 20, 2011 at 4:35 PM, satish patel satish...@hotmail.com wrote: Hey Thanks for that reply after add following option it works but the text output is totally different.. what its totally different is this dictionary problem ? -hmm /var/lib/asterisk/communicator -samprate 8000 In audio file its just: Hello satish this is test message 0: i started is it see no oil you did to less this tonight How many years have you spoken gibberish without knowing? Seriously, though, do you have a bit of an accent (compared to the pocketsphinx developers)? That's most likely the issue, I've seen weird stuff come out of STT apps that were running a British english dictionary and an American was speaking. Additionally, even the difference between accents in America (like between, for instance Maine/Vermont and Ohio, or even Boston and New York) can cause errors...Hell, I even have trouble sometimes deciphering what some people are saying here, and I've lived in almost every major accent area of the US ;-) -- _ -- 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] missed call notification
On Thu, Apr 21, 2011 at 12:26 PM, satish patel satish...@hotmail.comwrote:
Hi,
I am looking at http://www.theschmandts.org/blog/?p=28 to setup missed
call notification but i am having issue. following is my dialplan
[macro-stdexten]
exten = s,1,Dial(${ARG2})
exten = s,2,Goto(s-${DIALSTATUS},1); Jump
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(${ARG1},u) ; If unavailable,
send to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #,
return to start
exten = s-BUSY,1,Voicemail(${ARG1},b) ; If busy, send to
voicemail w/ busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they
press #, return to start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat
anything else as no answer
exten = a,1,VoicemailMain(${ARG1}) ; If they
press *, send the user into VoicemailMain
exten = h,1,System(/var/lib/asterisk/agi-bin/processcallemail.sh ${ARG3}
${CALLERID(num)} ${CALLERID(name)} ${DIALSTATUS} ${VMSTATUS}
${EXTEN})
[from-sip]
exten = _7[0123]XX,1,macro(stdexten,${EXTEN},sip/${EXTEN})
Following CLI output look like its not executing h extension in
macro-stdexten. But if i add h extension in [from-sip] it works! do you know
why ?
-- Executing [7207@from-sip:1] Macro(SIP/7101-000a,
stdexten,7207,sip/7207) in new stack
-- Executing [s@macro-stdexten:1] Dial(SIP/7101-000a,
sip/7207) in new stack
== Using SIP RTP CoS mark 5
-- Called 7207
-- SIP/7207-000b is ringing
== Spawn extension (macro-stdexten, s, 1) exited non-zero on
'SIP/7101-000a' in macro 'stdexten'
== Spawn extension (from-sip, 7207, 1) exited non-zero on
'SIP/7101-000a'
-- Executing [h@from-sip:1] Hangup(SIP/7101-000a, ) in new
stack
== Spawn extension (from-sip, h, 1) exited non-zero on
'SIP/7101-000a'
... google
http://www.voip-info.org/wiki/view/Asterisk+cmd+Macro
The Useful info was only a few lines from the beginning:
*'h' extension:* *If a macro executes a Dial() and the called party hangs
up, then the control passes to the 'h' extension of the calling context.
However, the 'h' extension is still needed inside the Macro context in case
of a command, application, or extension exiting non-zero - i.e. the user
hangs up in the middle of a Record() - in this case the 'h' extension of the
Macro context is used, not the 'h' extension of the calling context.)
Tilghman, May 2010: So Macro returns upon hangup to execute the h
extension in the original calling context, though even that is conditional,
based upon it having been broken for a long time.*
--
Sherwood McGowan
Telecommunications and VOIP Consultant
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Call Center Reporting
On Mon, Apr 18, 2011 at 5:49 AM, Steven Howes steve-li...@geekinter.netwrote: On 18 Apr 2011, at 11:06, bilal ghayyad wrote: I am using Asterisk for Call Center (so agents login, logout, ready, not ready, ... etc). To be able to have a good call center reporting, on what I have to depend? On the CDR of Asterisk or there is another way? Is there a good open source tool to be used for Asterisk call center reporting? http://www.google.com/search?q=asterisk+call+cener+reporting There are certainly some nice commercial ones. Can't comment on OSS stuff. But given it's for a call centre, I'd be tempted by the you don't get something for nothing approach. S -- I actually have implemented some very very good statistics by judicious use of the queue_log via odbc, running to a MySQL server, a few triggers, and then just wrote views that I can then filter on... Currently I have the following quickly accessible to me via running simple select queries (with date range constraints where needed): - Per Agent and Queue: - Number of calls taken - Can be grouped by day of month, day of week, or hour - Number and Percentage of calls: - answered - abandoned - timed out - exit via keypress - exit due to leavewhenempty constraints - broken down per queue - Can be grouped by day of month, day of week, or hour - Average, Minimum, and Maximum: - Call Duration - Per agent, queue, or both - grouped by day of month, day of week, or hour - Wait Time resulting in various results (abandon, exit with key, answer, etc...) - Per agent, queue, or both - grouped by day of month, day of week, or hour and that's just the stuff I did easily...I'm currently working on some more complicated statistics and the methods of generating those statistics... I hope to release my little stats project as an open source offering, and it will definitely be part of my open source PBX solution SamuPBX (Asterisk configuration web interface and supporting backend code, think FreePBX without all the overhead and flatfiles) -- _ -- 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] Call Center Reporting
If you want to know where I got the starting idea for the methods that I've developed, check out http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL and look at the section about using triggers -- _ -- 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, virendra bhati has invited you to open a Gmail account
On Mon, Apr 18, 2011 at 2:16 AM, virendra bhati virbh...@gmail.com wrote: I've been using Gmail and thought you might like to try it out. Here's an invitation to create an account. Just a guess, but I'm pretty sure that invitation will not work for everyone on the list ;-) -- _ -- 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] Call Center Reporting
On Mon, Apr 18, 2011 at 8:18 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote: If all the details you need to compile your reports can be found in existing databases (Asterisk's CDR database stores the details of calls; you may need to get user login/out events from a separate database), Logging the queue_log to MySQL and then setting up a trigger that inserts/updates data to other tables (such as something like agent_status and call_status), along with the CDR, will allow the OP to get pretty much everything they want. (*OP, if you need something substantially more than the stats I mentioned in my earlier post, definitely feel free to email me with details. That way, not only can I help you, but I can make the open source statistics solution I'm working on even better)* A hint: Do the whole thing -- or as much of it as it takes to prove to yourself that you're on the right track -- by hand first, entering all the queries yourself in the mysql prompt (or phpmyadmin), *before* you try to write a program to do it. You will save yourself much heartache that way. AJ, truer words have not been oft spoken! I'd also add that creating views helps if you have complex queries (just to shorten the query that has to be issued from the end program that gets written). -- _ -- 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 error logging
On Sat, Apr 16, 2011 at 6:05 PM, Jeremy Kister asterisk...@jeremykister.com wrote: bumping once before sending it to the tracker. Original Message Subject: [asterisk-users] sip error logging Date: Fri, 15 Apr 2011 03:39:23 -0400 I recently noticed that asterisk is not logging unknown sip connections. I'm not sure if I've broken something or if asterisk itself has been broken. the last entry I have is: /var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c: Registration from 'sip:22942@10.0.0.3' failed for '10.0.0.228:5060' - No matching peer found my logger.conf looks like: # grep -v '^;' /etc/asterisk/logger.conf [general] [logfiles] console = notice,warning,error,dtmf messages = notice,warning,error,verbose,dtmf,fax if i send 'options' or 'register' from a non-configured sip peer, i dont see anything in the log. am I missing something ? * i can replicate this behavior on 1.8.2.3 and 1.8.3.2 -- Jeremy Kister http://jeremy.kister.net./ This may sound like a stupid question, but what are your verbosity and debug levels set at currently? 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] sip error logging
On Sun, Apr 17, 2011 at 2:24 AM, Jeremy Kister asterisk...@jeremykister.com wrote: On 4/17/2011 3:16 AM, Sherwood McGowan wrote: This may sound like a stupid question, but what are your verbosity and debug levels set at currently? nope, thats exactly the type of thing i'm wondering if i'm missing :) but, i tried with verbose 3/debug 0 (which worked in 1.6), and i also tried with verbose 10/debug 10 before posting. no dice. Ah right on mate! Glad to see that you checked it *and* didn't mind being asked (after all, we're all IT/VOIP professionals, and we all know the first thing to ask is the simplest possible solution ;-] ) Cheers! -- _ -- 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] Variable inheritance with dialplan command Originate
On 4/12/2011 7:04 AM, Naomi Rosenberg wrote:
Hi Sherwood,
Thanks for helping me with this. The reply was indeed to you - I didn't think
you could use Dial on a channel that had been hung up, so I have learnt
something. However I'm still struggling with it I'm afraid. I've tried using
Dial and I'm finding that when the original channel is hung up it all seems
to stop working.
In the hope you might help me more, I've run your example as it is
(translated into .conf cos that's what we use here - feel free to reply in
ael) so I can show you the output. I'm finding it hangs just before the call
to Queue. I know Queue(2) works because when I dial 400 it works as expected.
[intern]
exten = 300,1,Goto(test-in,s,1) ; experiment
exten = 400,1,Queue(2) ; control
[test-in]
exten = s,1,Set(__referencenum=foo)
exten = s,n,Hangup();
exten = h,1,NoOp(The reference number is still here! ${referencenum})
exten = h,n,Dial(Local/123@staffcalls)
[staffcalls]
exten = 123,1,NoOp(reference number is STILL here ${referencenum})
exten = 123,n,Queue(2)
-- Executing [300@intern:1] Goto(SIP/200-0001, test-in,s,1) in
new stack
-- Goto (test-in,s,1)
-- Executing [s@test-in:1] Set(SIP/200-0001, __referencenum=foo)
in new stack
-- Executing [s@test-in:2] Hangup(SIP/200-0001, ) in new stack
== Spawn extension (test-in, s, 2) exited non-zero on 'SIP/200-0001'
-- Executing [h@test-in:1] NoOp(SIP/200-0001, The reference number
is still here! foo) in new stack
-- Executing [h@test-in:2] Dial(SIP/200-0001,
Local/123@staffcalls) in new stack
-- Called 123@staffcalls
== Spawn extension (test-in, h, 2) exited non-zero on 'SIP/200-0001'
-- Executing [123@staffcalls:1] NoOp(Local/123@staffcalls-c28c;2,
reference number is STILL here foo) in new stack
= then it just hangs here! ==
Naomi
I believe I made one mistake in my example, I don't use a call to Queue
in my local channel without a partner channel (the customer). I'll
revisit this later today when I have some time, I'll be glad to help you
if I can recall the right solution :)
--
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Carrier, ITSP, Call Center, and PBX Solutions Consultant
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Re: [asterisk-users] CEL Logging to MySQL - Please Test
On 4/12/2011 9:42 AM, Jonathan Penny wrote: I've recently finished an add-on module for CEL logging to MySQL, and it needs to be tested. The feature is being tracked at https://issues.asterisk.org/view.php?id=19058 And the patch is available at https://issues.asterisk.org/file_download.php?file_id=29110type=bug Thank You, -Jonathan Penny -- _ -- 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 Awesome! I'll try getting this into my latest development server -- 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] send voicemail to multiple emails
On 4/9/2011 11:56 PM, vip killa wrote: I've already taken the steps you described...issue i ran into was there is no variables passed to mailcmd only STDIN... as a result i have to extract variables from STDIN... On Fri, Apr 8, 2011 at 5:09 PM, Warren Selby wcse...@selbytech.com mailto:wcse...@selbytech.com wrote: On Fri, Apr 8, 2011 at 1:18 PM, vip killa vipki...@gmail.com mailto:vipki...@gmail.com wrote: That does not sound easy... besides these email addresses would be taken from a MySQL database. It's actually what you're going to end up doing, whether you do it on the MTA level or your code it into your script that you execute instead of sendmail -f. Currently, there is no way to natively have asterisk send one voicemail to multiple email addresses. What's probably going to work best for you since you seem to like program your own scripts (and I'm not talking an AGI here, I'm talking either pure bash, php, perl, or whichever you prefer), is to change the mailcmd= option inside voicemail.conf and replace it with a script of your own design. I'm not sure off the top of my head which variables are passed to the command, but you could always write a simple script that just outputs all arguments to see and go from there. My guess is you're going to at the least get the preconfigured email address and the contents of your emailsubject and emailbody options (both of which have the option of passing multiple useful variables). -- 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 -- _ -- 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 That's pretty much where you're at. What gets passed to STDIN is an email, it's not set up for use by a script. Remember, what you're doing is asking Asterisk to do something out of the ordinary, hence why having an email alias is typically the best solution. -- 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] Variable inheritance with dialplan command Originate
On 4/11/2011 5:15 AM, Naomi Rosenberg wrote:
Hi,
The reason I think Dial isn't appropriate is not to do with the database
call. Here's the wider context of the application I'm putting together:
Punter calls in, leaves a message, gets a reference number, hangs up. System
then initiates call to a queue of on-call staff and when one answers it plays
them the ref and the punter's message.
The Originate bit is when, after the punter's hung up, the system initiates
an outgoing call.
I've worked around the inheritance problem by using the reference number as
the extension, which being the primary key then allows me to retrieve the
rest of the data from the DB again once over the Originate hump.
Passing it all in the extension is an idea, but would not suit this case
since there is a lot of data and as the application develops the nature of
the data may change.
Naomi
I'm still not following why you think Dial is a bad idea. You're already
using a Local channel, which causes dialplan code to be executed upon
the start of the Local channel. Maybe you were replying to someone
else's post but hit reply on mine?
Your stated example in your email is pretty much EXACTLY what I'm
already accomplishing using Dial, Local Channels, and Variable
inheritance. Were it not for a Non-Disclosure Agreement that does not
allow me to share the specific code, I could show it to you and then
maybe you'd see what I'm trying to say.
Let's try a quickie example of what you're saying (I'm going to use AEL
this time, because typing same= over and over drives me nuts)
context inbound {
// punter calls in
_X. = {
// code for recording the message and database junk
// code returns a reference number to the caller
Set(__referencenum=foo); // this is the inherited variable
Hangup();
}
h = {
Noop(The reference number is still here! ${referencenum})
// Here is where we trigger the queue call to the staff
Dial(Local/123@staffcalls) ;
}
}
context staffcalls {
123 = {
Noop(reference number is STILL here ${referencenum});
// do your database lookup based on ${referencenum} here
Queue(staff) ; //obviously not a representation of your actual
queue request
}
}
the above example accomplishes what you're talking about, without
inheritance problems, and is working in a callcenter without issues.
--
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Carrier, ITSP, Call Center, and PBX Solutions Consultant
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Re: [asterisk-users] send voicemail to multiple emails
On 4/11/2011 12:30 PM, vip killa wrote: Would be so much simpler if mailcmd acted just like externnotify or externnotify only ran when a message was left but not when someone checks their voicemail... That's pretty much where you're at. What gets passed to STDIN is an email, it's not set up for use by a script. Remember, what you're doing is asking Asterisk to do something out of the ordinary, hence why having an email alias is typically the best solution. -- _ -- 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 I think you're confused about how it works. When someone leaves a voicemail, Asterisk sends a notification email, by piping an email to the defined mailcmd (by default, again, 'sendmail -t')...which is exactly what you're wishing it didmaybe you need to do some more reading? Why do you think it works in a different way? -- 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] send voicemail to multiple emails
On 4/11/2011 12:47 PM, vip killa wrote: Anyway, i figured out how to accomplish this using externnotify... In app_voicemail.c, in the function vm_execmain i commented out run_externnotify(vmu-context, vmu-mailbox, NULL); Now externnotify is called by asterisk only when there is a new message and not when someone checks their voicemail... -- _ -- 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 Ah, there was the problem, apparently somewhere along the way the fact that your complaint was with externnotify and not mailcmd got lost. -- 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] How to know the SIP status
On 4/11/2011 12:52 PM, virendra bhati wrote: Hi , As we see the SIP shatus on CLI with *sip show status How we get the status with phpagi function ? * -- - Thanks and regards Virendra Bhati +91-9172341457 Googling is your friend: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Command The API action Command runs an Asterisk CLI commanda command such as sip show status -- 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] Variable inheritance with dialplan command Originate
On 4/8/2011 4:57 AM, Naomi Rosenberg wrote:
Hi,
I would have thought that when spawning a channel using the Originate()
dialplan command, variables prefixed with two underscores would be preserved.
However this does not work in the following case.
Dialplan code:
[intern]
exten = 200,1,Set(__myvar=foo)
exten = 200,n,Originate(Local/123@test_orig,exten,dummy)
[test_orig]
exten = 123,1,NoOp(${myvar})
exten = 123,n,Hangup()
[dummy]
/end dialplan code.
Console output:
-- Executing [200@intern:1] Set(SIP/200-0018, __myvar=foo) in
new stack
-- Executing [200@intern:2] Originate(SIP/200-0018,
Local/123@test_orig,exten,dummy) in new stack
-- Executing [123@test_orig:1] NoOp(Local/123@test_orig-cbab;2, ) in
new stack
-- Executing [123@test_orig:2] Hangup(Local/123@test_orig-cbab;2, )
in new stack
/end console output.
This is in Asterisk 1.8.3.
Is this expected behaviour or a bug, or am I just confused? I would
appreciate your thoughts on the matter.
Thank you,
Naomi
I believe that it's expected behavior because you're not creating a
child channel, you're originating a different set. Try using Dial
instead of Originate, and you'll get the inheritance behavior you expected.
--
Sherwood McGowan sherwood.mcgo...@gmail.com
Carrier, ITSP, Call Center, and PBX Solutions Consultant
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Re: [asterisk-users] Variable inheritance with dialplan command Originate
On 4/8/2011 10:57 AM, Naomi Rosenberg wrote: Thanks. That's as I thought (feared). Dial is not an option in this case but I have come up with a workaround involving using a reference number as the extension and then doing a database call. Not pretty but it works! Naomi I'm not sure why Dial wouldn't work...I use Dial all the time for triggering Local channels that perform database calls all the time -- 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] Variable inheritance with dialplan command Originate
On 4/8/2011 11:05 AM, Jim Dickenson wrote:
Another option is to pass the information in the extension. At times I have
an extension like
_[s][o][m][e]-[e][x][a][m][p][l][e].
And call it like some-example:info1:info2 and use cut to extract the info1
and info2 values. Not real pretty but as this is computer generated calls it
gets the job done.
Still not sure why you guys need this...Here's my example
[firstleg]
exten = 200,1,Set(__myvar=foo) ; Don't forget you don't want quotes!)
exten = 200,n,Dial(Local/123@test_orig)
[test_orig]
exten = 123,1,Noop(${myvar})
same = n,Set(dbtest=${ODBC_TESTQUERY(myvar)})
--
Sherwood McGowan sherwood.mcgo...@gmail.com
Carrier, ITSP, Call Center, and PBX Solutions Consultant
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Re: [asterisk-users] send voicemail to multiple emails
On 4/8/2011 1:13 PM, vip killa wrote: Is there a way for asterisk's voicemail to send an email (including voicemail attachment) to multiple email addresses? -- _ -- 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 The easiest way would be to set up an alias in your MTA configuration. That way, you could configure the mailbox for the alias email address and copies would be sent to all addresses in the alias definition. -- 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] send voicemail to multiple emails
On 4/8/2011 1:18 PM, vip killa wrote: That does not sound easy... besides these email addresses would be taken from a MySQL database. The easiest way would be to set up an alias in your MTA configuration. That way, you could configure the mailbox for the alias email address and copies would be sent to all addresses in the alias definition. -- _ -- 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 Easy for you and easiest to configure are two different things. Aliasing email addresses to multiple addresses is not a problem for me, so I shared what I know. You could also just replace the mailcmd (usually sendmail -t) with a script that you wrote (or even application, if you care to compile something) that would take care of the functionality you wish. If you don't think that's easy, just ignore my message, we're not here to compare skill levels. -- 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] send voicemail to multiple emails
On 4/8/2011 1:20 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Sherwood McGowan Sent: Friday, April 08, 2011 1:16 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] send voicemail to multiple emails On 4/8/2011 1:13 PM, vip killa wrote: Is there a way for asterisk's voicemail to send an email (including voicemail attachment) to multiple email addresses? -- _ -- 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 The easiest way would be to set up an alias in your MTA configuration. That way, you could configure the mailbox for the alias email address and copies would be sent to all addresses in the alias definition. -- Sherwood McGowan sherwood.mcgo...@gmail.com [Danny Nicholas] That is a grand suggestion - as much as I like Asterisk, it is always easier to let Linux do the grunt work when applicable. -- _ -- 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 Thanks Danny, it's the solution I've used many many times :) -- 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] Call duration problem or maybe calls not hanging up problem
Very weird mate...I would have replied sooner, but in reality there's a LOT of troubleshooting to be done and it would require working with your provider. It sounds like (if you're sending a bye when your calls disconnect) you never receive an actual 200 OK stating the call is picked up and so your system is sending a CANCEL ? Just spitballing here -- _ -- 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] No ringback even though progressinband=yes is set
On 4/7/2011 11:02 AM, Douglas Mortensen wrote: Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository: asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9 Any help would be greatly appreciated! :-) - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 . -- _ -- 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 If you are referring to a ringback tone when they first dial your system, meaning that they immediately hear your IVR when they dial your PBX's number, it's because that's how it's supposed to work. Unless you tell your PBX to use the Ringing() app and wait for a period of time, Asterisk normally picks up at the beginning of the IVR (since the first thing you have to do to send audio via Background or Playback is issue the command Answer() to start sending actual audio. (Note: The Ringing app just signals RINGING to the remote party) -- 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] No ringback even though progressinband=yes is set
On 4/7/2011 4:54 PM, Douglas Mortensen wrote: I have inbound calls going directly to a ring group. When callers call in, they (the callers) hear complete silence even though the phones that are part of the ring group ARE ringing properly. Employees can answer the calls when their phones ring, and everything works fine. The problem is simply that the external caller never hears any ringing. Even if the SIP phones in the ring group ring for 5 rings, it is total silence even though there is ringing going on inside of the office. I'm pretty sure it is a ringback issue. I'm going to try to turn on SIP debugging see what I can figure out that way. I do appreciate your help. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 If you're using an interface (I believe you said AsteriskNOW), you might want to check the Dial Options...Make sure that 'r' is one of the options. The reason you're not hearing ringing is probably due to Asterisk not sending a RINGING signal. If you have 'r' defined in the dial options in your interface, then AsteriskNOW is probably using a Dial command that is NOT using your global dial options. -- 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] Iptables configuration to handle brute, force registrations?
On 4/5/2011 4:38 PM, Paul Dugas wrote: First, this appears to be working for me though I'm not 100% sure of that and cannot guarantee it will for you in any way, shape or form. With the lawyering out of the way... I've seen fail2ban allow more than 500 failed SIP login attempts in under 30 seconds before adding an iptables rule to block the attacker. Likely I have it configured wrong but lately, I've been tinkering with iptables rules using the recent module as another layer of defense. Relevant lines from /etc/sysconfig/iptables on my CENTOS/Asterisk machine below... -A RH-Firewall-1-INPUT -p udp --dport 5060 -m state --state NEW -m recent --set --name SIP -A RH-Firewall-1-INPUT -p udp --dport 5060 -m state --state NEW -m recent --rcheck --name SIP --seconds 600 --hitcount 20 --rttl -j DROP -A RH-Firewall-1-INPUT -p udp --dport 5060 -m state --state NEW -m recent --rcheck --name SIP --seconds 300 --hitcount 10 --rttl -j DROP -A RH-Firewall-1-INPUT -p udp --dport 5060 -m state --state NEW -m recent --rcheck --name SIP --seconds 180 --hitcount 5 --rttl -j DROP -A RH-Firewall-1-INPUT -p udp --dport 5060 -m state --state NEW -m recent --rcheck --name SIP --seconds 60 --hitcount 3 --rttl -j DROP -A RH-Firewall-1-INPUT -p udp --dport 5060 -j ACCEPT This blocks the attacker when too many new SIP connections happen in too short a period of time. I think fail2ban will now never sees enough failed logins to fire off a response. $0.02 That was completely worth the $0.02, here's a nickel keep the change! ;-) Cheers mate, thanks for sharing with the community :) -- 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-users Digest, Vol 81, Issue 12
On 4/5/2011 2:45 PM, Bill Michaelson wrote: On 04/05/2011 03:06 PM, asterisk-users-requ...@lists.digium.com wrote: Message: 12 Date: Tue, 5 Apr 2011 13:36:21 -0500 From: Sherwood McGowan sherwood.mcgo...@gmail.com Subject: Re: [asterisk-users] Iptables configuration to handle brute, force registrations? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Bill Michaelson b...@cosi.com Message-ID: banlktimqrbfmqpoinrphr_rjekolbwp...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 On Tue, Apr 5, 2011 at 1:31 PM, Bill Michaelson b...@cosi.com wrote: fail2ban might be good for this. I think you missed the point, which is reducing the need for an external application that searches logs in order to determine whether or not to block an IP. Why run fail2ban and add overhead when you can just do the same thing with iptables itself? I apologize for jumping into the middle without reading the beginning of the discussion in which this central requirement to avoid an external application was stated, as I now infer from Mr. McGowan. Sorry for missing the point. I'll have to read up on fail2ban also. I thought it monitored the tails of logs. I did not know that it searched them. My intent was to suggest using an established tool that would consolidate the IP blocking and unblocking function for all ports into a single application without imposing additional maintenance overhead of new code for this purpose. Obviously, I'm not seeing the big picture. Sorry for my myopic comments and for cluttering the list. I won't make the mistake of offering worthless contributions in the future. -- _ -- 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 *grabs a bucket of water*. My my my, I think I need some hipwaders.Was that a Caterpillar brand roadgrader you were using to spread that sarcasm on so thick? My reply to your statement about fail2ban was not intended to be rude/sarcastic/mean...After being a complete jackass on this list a few times in the last 6 months, I realized that I was, and I've kept my flamethrower in storage as of late... I wasn't trying to chastise you, or whatever you may have thought from my reply. My reply may have not necessarily been constructive either, but give a recovering threadflamer a little credit, there wasn't really even a spark in my email... Slainte all, I'm off to bed -- 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] Call recording - methodology
On Wed, Apr 6, 2011 at 5:54 AM, Silver Thorne szilvertho...@gmail.comwrote: Hello Everyone; I am looking for a solution to record calls that come into our Asterisk server. I am hoping for something that is easy to use - however, if I have to modify it to make it easier to use, I do not mind. Does anyone know of any opensource or otherwise solutions out there that I can try out? Thanks much. Glen -- _ -- 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 Hrm Try googling MixMonitorAsterisk has built in call recording -- _ -- 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] Iptables configuration to handle brute, force registrations?
On Tue, Apr 5, 2011 at 1:31 PM, Bill Michaelson b...@cosi.com wrote: fail2ban might be good for this. I think you missed the point, which is reducing the need for an external application that searches logs in order to determine whether or not to block an IP. Why run fail2ban and add overhead when you can just do the same thing with iptables itself? -- _ -- 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] Iptables configuration to handle brute, force registrations?
On 4/5/2011 2:11 PM, Steve Edwards wrote: On Tue, 5 Apr 2011, Sherwood McGowan wrote: Why run fail2ban and add overhead when you can just do the same thing with iptables itself? Because it's not the same? The iptables approach is great because it is 'light-weight' and it should already 'be there.' Also, it can react quicker because it doesn't have to read log files to make a decision. The 'downside' of the iptables approach is that the blocks go away when iptables is reloaded -- like when the host is restarted. Probably not an issue with Gordon since his hosts stay up for years. I'm thinking the iptables approach supplemented with a script to periodically save the block list to disk would allow persistent blocks as well as letting you accumulating blocks between all your hosts. Which would still be much 'lighter' than fail2ban. Agreed on all points Steve. I've already implemented an auto save function, to workaround the drawback you mentioned. Are there possibly other drawbacks that I'm not seeing/remembering? I've been running an iptables based setup for some time, never really jumped into the fail2ban wagon -- 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] 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] 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 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 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
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] 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
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] Variable. AMI and dialplan
On 3/28/2011 7:41 AM, magnu...@inputinterior.se wrote: Hi! Guess I am doing something totally wrong here: Some smart person could maybe plz tell me what. From AMI, I set a variable Action: Setvar\r\nVariable:x\r\n\Value: 5\r\n\r\n From dialplan i can “access” the variable “x” and see the value “5” From dialplan i modify “x” to “8”. But from AMI i still se “x” as “5” not “8”. /Magnus Maybe you need to perform a GetVar to read the new value of that channel variable -- 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] Variable. AMI and dialplan
Don't know then, that's all I've got far ya today mate, sorry On 3/28/2011 8:18 AM, magnu...@inputinterior.se wrote: I did use Action: Getvar when i read it back in AMI. On 3/28/2011 7:41 AM, magnu...@inputinterior.se wrote: Hi! Guess I am doing something totally wrong here: Some smart person could maybe plz tell me what. From AMI, I set a variable Action: Setvar\r\nVariable:x\r\n\Value: 5\r\n\r\n From dialplan i can “access” the variable “x” and see the value “5” From dialplan i modify “x” to “8”. But from AMI i still se “x” as “5” not “8”. /Magnus Maybe you need to perform a GetVar to read the new value of that channel variable -- 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] 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] AMI redirect from Queue to MeetMe
On 3/28/2011 10:02 AM, Jim Dickenson wrote:
I would be surprised that you did not always hang up the second
channel you are redirecting. Once you transfer one leg there is
nothing connected to the second leg so it goes away, I would think.
What we do is remember the agent number, transfer the caller, and then
setup a call to the agent and meetme room.
More or less like:
Action: Redirect
Channel: SIP/GXP280_18-0001
Exten: do_meetme601MyID
Context: cfmc_cdi_private
Priority: 1
ActionID: MeetMe
Async: true
Action: Originate
Channel: Agent/1001
Exten: do_meetme601MyID2
Context: cfmc_cdi_private
Priority: 1
ActionID: DirectMeet
Async: true
exten = _do_meetme.,1,UserEvent(BeforeMeetMe,Info:${EXTEN:9}
${UNIQUEID} ${CHANNEL})
exten = _do_meetme.,n,Answer()
exten = _do_meetme.,n,Set(CfMC_RoomToUse=${EXTEN:9:3})
exten = _do_meetme.,n,Set(CfMC_CurrentID=${EXTEN:12})
exten = _do_meetme.,n,Set(MEETME_MOH_CLASS=meetme-music)
exten = _do_meetme.,n,MeetMe(${CfMC_RoomToUse},CMpqx1)
exten =
_do_meetme.,n,UserEvent(AfterMeetMe,ActionID:${CfMC_CurrentID}
Room:${CfMC_RoomToUse} ${UNIQUEID} ${CHANNEL})
exten = _do_meetme.,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 28, 2011, at 1:23 AM, Deka, Rajib IN MAA SL wrote:
Hello List,
I have scenario as follows,
1. A call comes to queue.
2. Available agent will answer the call.
3. BridgeEvent wil be generated in AMI with channel1 and channel2.
4. Parse channel1 and channel two from the event and redirect them
to a meetme room,
Dialplan,
Exten = 1234,1,MeetMe(1234,1dq)
But sometime it works and sometime one leg gets disconnected after
redirection. Is it a bug to asterisk-1.6.2.13 ?
Regards,
*Rajib Deka*
SIEMENS Ltd.
RobertV Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com http://www.siemens.com/
Mob: +91-9176780669| E-Mail: rajib.d...@siemens.com
mailto:rajib.d...@siemens.com
Important notice: This e-mail and any attachment there to contains
corporate proprietary information. If you have received it by
mistake, please notify us immediately by reply e-mail and delete this
e-mail and its attachments from your system.
Thank You.
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You could also use the Extra options (Channel, context, extension,
priority) to transfer BOTH legs
--
Sherwood McGowan sherwood.mcgo...@gmail.com
Carrier, ITSP, Call Center, and PBX Solutions Consultant
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Re: [asterisk-users] s extension not working
Uhm That's because you're being passed 7527 as the extension, so it won't match s On 3/28/2011 11:38 AM, satish patel wrote: If i use 's' then i got following error. This scenario is back to back asterisk connected on PRI line (T1). for testing purpose i calling from one asterisk to other and i want to land call on 's' extension. shirley*CLI -- Extension '7527' in context 'from-pstn' from '7623' does not exist. Rejecting call on channel 0/1, span 1 If i use _XXX then it working with following output. shirley*CLI -- Accepting call from '7623' to '7527' on channel 0/1, span 1 -- Executing [7527@from-pstn:1] Answer(DAHDI/i1/7623-10, ) in new stack -- Executing [7527@from-pstn:2] Playback(DAHDI/i1/7623-10, hello-world) in new stack -- DAHDI/i1/7623-10 Playing 'hello-world.ulaw' (language 'en') -- Executing [7527@from-pstn:3] Hangup(DAHDI/i1/7623-10, ) in new stack == Spawn extension (from-pstn, 7527, 3) exited non-zero on 'DAHDI/i1/7623-10' -- Hungup 'DAHDI/i1/7623-10' From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 28 Mar 2011 11:08:57 -0500 Subject: Re: [asterisk-users] s extension not working *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *satish patel *Sent:* Monday, March 28, 2011 11:04 AM *To:* asterisk-users *Subject:* [asterisk-users] s extension not working Hey Guys! I have asterisk 1.8.x and somehow my 's' extension not picking up any incoming calls.. Not working [from-pstn] exten = s,1,Answer() same = n,Playback(hello-world) same = n,Hangup() Working... [from-pstn] exten = _,1,Answer() same = n,Playback(hello-world) same = n,Hangup() -S Ok Satish. I assume sip.conf or dahdi.conf has a context of from-pstn. The key to actually solving this will be for you to give us say 10 lines of CLI output. -- _ -- 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 -- 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] s extension not working
On 3/28/2011 1:33 PM, satish patel wrote: @Sherwood, I was also thinking about that But then how 's' extension match any unknown number ? Like when call coming from PSTN then how IVR picked up...? -Satish The 's' extension does not match anything other than 's'. If your sip registrations are configured without a trailing /DIDNUMBER, it gets sent to the 's' extension on your default context. However, if you want to match *any* number, you'd want '_X.', which matches any number. -- 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] pbx.c: We were unable to say the number
On Sun, Mar 27, 2011 at 2:50 PM, Mohammad Khan beepl...@gmail.com wrote:
Here is the dialplan in macro:
exten = s,n,SayNumber($[${ARG1} % 100])
when 662 was passed as ARG1, I had the following at log:
WARNING[15217] pbx.c: We were unable to say the number 62, is it too large?
Do you see any odd in my dialplan?
662 % 100 = 66.2, not 62. It seems to me that there's more going on
here..Maybe Asterisk is being confused by actually getting 66.2? I'm not
readily able to look into the source, but I *think* that Asterisk (or at
least, SayNumber) cannot handle a number with a decimal point, but please
don't take that as gospel.
If Tilghman's question doesn't result in a fix, the next thing I'd say is to
check and make sure that a floating point number can be supplied as an
argument to SayNumber.
Also make sure you're logging verbose, debug, error, and warning messages
into a logfile, bump the verbosity and debug up to 5, and then run another
test call that will result in the number(s) you're testing. Next, send off a
larger amount of the content from the logfile, the single WARNING line is
not enough for anything more than a blind guess. For instance, it would be
REALLY fantastic is you would send the log line that displays SayNumber
actually being executed, like ( *[DATETIME] VERBOSE[23609] pbx.c: --
Executing [s@contextname:priority] SayNumber(CHANNELNAME, 66.2) in new
stack *)... It would be even MORE fantastic if you included almost ALL of
that call's log output, but at the very LEAST there should be around 5
lines, starting from the verbose output for execution of SayNumber. That
way, we don't just get the warning message you're complaining about, but the
EXACT executions and messages outputted leading up to the warning message.
But then again, you could continue to do essentially the same thing and hope
for different results...
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Re: [asterisk-users] pbx.c: We were unable to say the number
Oh crap, you're right, my bad. Yes, I also agree, it's most probably the
language and/or missing files
On Sun, Mar 27, 2011 at 4:30 PM, Jeff LaCoursiere j...@sunfone.com wrote:
On Sun, 2011-03-27 at 16:14 -0500, Sherwood McGowan wrote:
On Sun, Mar 27, 2011 at 2:50 PM, Mohammad Khan beepl...@gmail.com
wrote:
Here is the dialplan in macro:
exten = s,n,SayNumber($[${ARG1} % 100])
when 662 was passed as ARG1, I had the following at log:
WARNING[15217] pbx.c: We were unable to say the number 62, is
it too large?
Do you see any odd in my dialplan?
662 % 100 = 66.2, not 62. It seems to me that there's more going on
here..Maybe Asterisk is being confused by actually getting 66.2? I'm
not readily able to look into the source, but I think that Asterisk
(or at least, SayNumber) cannot handle a number with a decimal point,
but please don't take that as gospel.
'%' is 'modulus', and 62 is the correct result. I am betting it is the
language setting, and missing audio files.
j
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Re: [asterisk-users] pbx.c: We were unable to say the number
On Fri, Mar 25, 2011 at 11:15 PM, Mohammad Khan beepl...@gmail.com wrote: Hello, Occasionally, I get the following warning in my asterisk log, pbx.c: We were unable to say the number [n], is it too large? n is two or one digit number, which doesn't look like large to me! Could anybody please tell more about this warning, like in what scenario I may have this warning. Thanks, Mohammad Please post the relevant context that is being executed, that'll give us not only the actual application, but more info as to how it's being passed. -- _ -- 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] pbx.c: We were unable to say the number
Again, the relevant dialplan code is important. It is quite possible that there's an issue with the dialplan code that you (as the person who's dealing with the issue) may have missed. It happens all the time. On Sat, Mar 26, 2011 at 1:25 PM, Mohammad Khan beepl...@gmail.com wrote: I am using asterisk 1.4.38 I am getting this warning occasionally when executing SayNumber in a macro with argument which is less than 100. On Sat, Mar 26, 2011 at 11:03 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On Fri, Mar 25, 2011 at 11:15 PM, Mohammad Khan beepl...@gmail.comwrote: Hello, Occasionally, I get the following warning in my asterisk log, pbx.c: We were unable to say the number [n], is it too large? n is two or one digit number, which doesn't look like large to me! Could anybody please tell more about this warning, like in what scenario I may have this warning. Thanks, Mohammad Please post the relevant context that is being executed, that'll give us not only the actual application, but more info as to how it's being passed. -- _ -- 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] White papers or success cases to convince a customer?
On Fri, Mar 25, 2011 at 6:05 PM, Carlos Chavez cur...@telecomabmex.comwrote: Can anyone recommend some White Papers or Success Cases that we can use to ease the mind of a customer that has not heard much about Asterisk? All they know is Avaya at this point. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 Hopefully someone can point you to some papers, but if you end up just needing someone to write up a testimonial about how Asterisk has been a successful part of several (20+) projects, contact me offlist and I'll write one up. I've used Asterisk in projects ranging from very small business PBX's all the way up to large VOIP telephone service providers. -- _ -- 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] Forwarding XXXX to XXXX prevented.
Well, first, I'd say more information is needed I see you're using a Local channel construct, is it pointing to a valid context and extension? Is there any more debugging information you can provide? It seems there's something missing here, if I was debugging the issue to find a solution, I'd be digging up a lot more info, but we're not local to the problem, we have to rely on you the poster. On Thu, Mar 24, 2011 at 12:21 PM, Ernie Dunbar maill...@lightspeed.cawrote: So... no solution to this problem? It does depend on how you set up the call forwarding on asterisk and sometimes when the ATA sends the forwarding call to the Voip provider server it has nothing to do with it which causes a problem. if you disable call forwarding remotely see if that works also. its a tricky situation. On Wed, 2011-03-23 at 16:18 -0700, Ernie Dunbar wrote: I have a Linksys 2102 ATA here that does call forwarding internally with the *72 code, however our Asterisk 1.6.2.17 server doesn't forward the call properly. This is what shows up in the console when an incoming call is made while the ATA is call-forwarded: -- Called Username -- Got SIP response 302 Moved Temporarily back from XX.XXX.XX.XXX -- Now forwarding DAHDI/1-1 to 'Local/12505551234@vancouver' (thanks to SIP/Username-0045) -- Forwarding DAHDI/1-1 to 'Local/12505551234@vancouver' prevented. == Everyone is busy/congested at this time (1:1/0/0) The SIP configuration allows call forwarding (cancallforward=yes), so I'm at a loss as to what is preventing the forwarding. It's not like Asterisk is very specific about that. -- _ -- 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
Re: [asterisk-users] Multi-Tenant Hosted PBX system with Reseller functionality
Thanks John Bower and ITSPTEC.COM, you've made it easy for me to not feel bad about never using your products... On Mon, Mar 21, 2011 at 7:45 PM, Juan hardwareven...@gmail.com wrote: damn, advertisements everywhere, also in non commercial mailing lists... ITSPTEC.COM seems don't understand what a NON-COMMERCIAL DISCUSSION is about I will never buy anything from people like you who don't seems to understand so basic things @itsptec.com should be blacklisted... On Mon, 21 Mar 2011 20:38:10 -0300, john.bo...@itsptec.com wrote: We are glad to announce that ITSPtec now offers a complete ITSP system for Asterisk with powerful routing engine, billing System- including invoicing, configuration, phone auto-provisioning and tones of other features. For Asterisk, we offer a Multi-Tenant Hosted PBX system with Reseller functionality that can be used by ITSP’s to provide HostedPBX services with reseller capability. For more information, please visit us at http://www.itsptec.com 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 -- _ -- 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] Multi-Tenant Hosted PBX system with Reseller functionality
Oh my... On Mon, Mar 21, 2011 at 8:09 PM, Outback Dingo outbackdi...@gmail.comwrote: Even worse... now it smells of a scam Registered through: GoDaddy.com, Inc. (http://www.godaddy.com) Domain Name: ITSPTEC.COM Created on: 27-Jan-11 Expires on: 27-Jan-12 Last Updated on: 27-Jan-11 On Mon, Mar 21, 2011 at 9:06 PM, Outback Dingo outbackdi...@gmail.comwrote: great way to kill sales for your company idiot.! On Mon, Mar 21, 2011 at 7:38 PM, john.bo...@itsptec.com wrote: We are glad to announce that ITSPtec now offers a complete ITSP system for Asterisk with powerful routing engine, billing System- including invoicing, configuration, phone auto-provisioning and tones of other features. For Asterisk, we offer a Multi-Tenant Hosted PBX system with Reseller functionality that can be used by ITSP’s to provide HostedPBX services with reseller capability. For more information, please visit us at http://www.itsptec.com 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 -- _ -- 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] Getting the missed calls using Asterisk Manager
Missed calls would most easily be found by looking in the Call Detail Records (CDRs)... 2011/3/17 Octavian Rasnita orasn...@gmail.com Hi, Is it possible to get the number of missed calls for a certain phone number using Asterisk Manager? If yes, please tell me how, or tell me where to look. Also, is it possible to find who were those who dialed? I have read the section of Asterisk manual about the Asterisk Manager, and I got the list of all the commands using the help command, but I couldn't find the command for getting the missed calls. Thank you. --Octavian -- _ -- 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] Loudness of recorded wav-audio
You could always just use sox to adjust the levels -- _ -- 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] Prepaid Billing other than A2Billing
If you're in the market for a custom solution for whatever reason, there's more than a few of us who can write a custom prepaid solution. I've done about 7 so far personally and I know there's more like me out there On Sat, Mar 5, 2011 at 11:26 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; Any one advise for open source prepaid billing other than A2Billing that can work with Asterisk and it is rich by features (for large business)? 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] SIP Hardphone that works well with asterisk
I'm VERY partial to Aastra's devices. Seriously, they don't take as long to boot as Polycoms, they're relatively inexpensive but are not CHEAP (like a certain brand beginning with a G, in my opinion), they have decent web interfaces (also unlike the unnamed brand I non-mentioned a moment ago), solid features, and have good expansion modules as well. In almost every case where a client ends up choosing Aastra as the brand to buy after reading my shortlist of brands/models I suggest, I've had less overall issues/complaints about the phones themselves. Even Polycom based clients/locations have more issues/complaints/annoyances they wish to bring to my attention than the Aastra based ones. I do have a complaint about Aastra though...Because of my client's happiness with the brand, and because I personally think they're worth suggesting, I spoke with Aastra about becoming an authorized reseller...filled out the paperwork, scanned it and emailed it to the rep I was working with..and never heard another word...For a phone device company to never get back to a telecom consultant who wanted to not only put their brand at the top of the shortlist, but wanted to push the brand as their preferred brand to use..I mean come on, seriously, free advertising and purchase referrals? Why wouldn't you just file the paperwork and reply to the completed application with a thank you, you are now an authorized reseller or even just a thanks, but you don't meet the projected sales numbers we wish from our resellers? Anyway, Aastra IS great in my book... Cheers all! Sherwood I'm the Mick and that's my $0.05..keep the change McGowan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variables losing their value????
Apologies, using two underscores (I retested) did not cause the error
On Sat, Feb 12, 2011 at 1:42 AM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
Alrighty Gents, let's see if any of you have encountered this
one...Variables losing their value...I'm setting a variable with four
underscores (used to be two, had same issue) so it can be inherited by child
channels, and then the next line in the dialplan I use it but it appears to
be empty...I've googled and found nothing stating this kind of weirdness..
Asterisk 1.8.2.2 (upgrading to 1.8.2.3 shortly)
dialplan:
[menu.main]
exten = s,1,Set(recfile=${FILTER(0-9,${UNIQUEID})});
exten = s,n,Set(logfile=${recfile}) ;
The log output:
-- Executing [s...@menu.main:1] Set(SIP/-,
recfile=12974953060) in new stack
-- Executing [s...@menu.main:2] Set(SIP/-, logfile=)
in new stack
Anybody have thoughts?
Thanks,
S McGowan
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[asterisk-users] Variables losing their value????
Alrighty Gents, let's see if any of you have encountered this
one...Variables losing their value...I'm setting a variable with four
underscores (used to be two, had same issue) so it can be inherited by child
channels, and then the next line in the dialplan I use it but it appears to
be empty...I've googled and found nothing stating this kind of weirdness..
Asterisk 1.8.2.2 (upgrading to 1.8.2.3 shortly)
dialplan:
[menu.main]
exten = s,1,Set(recfile=${FILTER(0-9,${UNIQUEID})});
exten = s,n,Set(logfile=${recfile}) ;
The log output:
-- Executing [s...@menu.main:1] Set(SIP/-,
recfile=12974953060) in new stack
-- Executing [s...@menu.main:2] Set(SIP/-, logfile=) in
new stack
Anybody have thoughts?
Thanks,
S McGowan
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Re: [asterisk-users] Call Recording audio file quality query
Tilghman, When you say reformat the audio, do you mean sample rate and bits per sample, etc...or do you mean the format in which each packet of data is structured ? I just want to make sure I know which one I'd be dealing with if recording a call that was using one of the higher quality codecs that was metioned earlier. I *think* you mean just the structure version of the format options I presented, because for example: Microsoft PCM (wav) files can be of varying quality levels (192Khz, 256Khz..8bit 16 bit 24...32)..This is true (as you know, I'm more than sure) of almost every audio file format... So, is it Structure of data/packets or sample rate, bitrate, etc' ? Thanks mate! -- _ -- 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] Call Recording audio file quality query
On Wed, Feb 9, 2011 at 12:31 PM, Tilghman Lesher tilgh...@meg.abyt.eswrote: On Wednesday 09 February 2011 03:50:51 Sherwood McGowan wrote: Tilghman, When you say reformat the audio, do you mean sample rate and bits per sample, etc...or do you mean the format in which each packet of data is structured ? I just want to make sure I know which one I'd be dealing with if recording a call that was using one of the higher quality codecs that was metioned earlier. I *think* you mean just the structure version of the format options I presented, because for example: Microsoft PCM (wav) files can be of varying quality levels (192Khz, 256Khz..8bit 16 bit 24...32)..This is true (as you know, I'm more than sure) of almost every audio file format... So, is it Structure of data/packets or sample rate, bitrate, etc' ? That would be structure of data stored in the file. At the point where the file format comes into play, the samples are already in their final stage of computation. The only thing that remains is how the samples are wrapped for storage. -- Tilghman thanks for confirming! -- _ -- 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] Call Recording audio file quality query
On Tue, Feb 8, 2011 at 5:09 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi We're getting requests coming in for higher quality audio in our call recordings. We currently use MixMonitor and everything is being saved in it's native 8000Hz, 16 bit wav format. I have seen information on using Monitor and specifying a conversion to mp3 when the call ends and the 2 channels get mixed but surely the 2 channels are already saved as 16bit 8000Hz wav files so the quality is lost already? Is there any way of making high quality recordings of call content? Have you ever heard of the saying You can't polish a turd ? It doesn't matter if you have an app capable of recording 196Khz 24bit recordings (or capable of upsampling to that sample rate)...if the call itself is native at 8Khz 16bit, you'd just be making a bigger recording file with no literal improvement in quality. You can't create more samples of audio from nothing. it's like taking a new box of, say, 50 paperclips... Now, go get an empty box that says it contained 250 paperclips when it was purchased... Now, throw all 50 paperclips from the little box into the big box marked 250..now, imagine REALLY REALLY hard that you think you can perceive about 5 more paperclips somewhere all mixed up in the jumble...(Extrapolation) that, my friend, is an over simplified metaphor, but in essence it's close enough to get the point across.. Sorry bud :( If you don't believe me, I can refer you to my old audio production school ;-D ) Slainte! the Mick -- _ -- 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] Call Recording audio file quality query
That answer was pretty much what I was expecting. Just wanted to make sure. Glad to be of service :D -- _ -- 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] Call Recording audio file quality query
On Tue, Feb 8, 2011 at 6:01 AM, fai...@vopium.com wrote: But if you are getting calls all the way on VoIP then you can have calls in HD audio using HD audio codec on all locations (Server and Client). In that case you either need use some available 3rd party solution which uses packet capturing to trace the calls and record call using packet capture and assembling regardless of server as asterisk still will not be able to record call in HD but some other switches like FreeSWITCH can do it or you need to write your own app like it. It's not difficult at all to perform what you're referring to..If you have the hardware... A simple way is to have a port on your main network switch/router that will firehose the traffic the device interacts with In case someone reading this doesn't know, I'm talking about having a port that just makes a copy of EVERY PACKET that the device sees and sends those copies out over the port that you've set up for the purpose..It just GUSHES data over that port...like a firehose just gushes out all the water it possibly can... LOL Anyway, once your data is being mirrored over that firehose, send it to a dedicated recording server...all it has to do is find the signaling packets for each call and then just dump the payload from the RTP. It'll come out exactly as it was transported within RTP...in the codec the call set up I may be wrong, but I'm fairly sure that Asterisk can write a filetype for almost any of it's codecs...I know it can READ audio files that are encoded in GSM, uLaw (ul), aLaw (al), G726 and G729 formats (.g729, g.726)...etc... If the DECoding portion is there, there's almost GOT to be the enCOding functionality... -- _ -- 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] Call Recording audio file quality query
yep..that would be what i said, using the nifty slang my peeps use in the datacenters I just wanted to be cool like them...*hangs head*... great...now I gotta transfer to another school... LOL, have a good one mate! On Tue, Feb 8, 2011 at 7:23 AM, fai...@vopium.com wrote: Yes. The technology need to be used on LAN switches is port mirroring or line tapping -Original Message- From: Sherwood McGowan sherwood.mcgo...@gmail.com Sent: Tuesday, February 8, 2011 7:34am To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Call Recording audio file quality query On Tue, Feb 8, 2011 at 6:01 AM, fai...@vopium.com wrote: But if you are getting calls all the way on VoIP then you can have calls in HD audio using HD audio codec on all locations (Server and Client). In that case you either need use some available 3rd party solution which uses packet capturing to trace the calls and record call using packet capture and assembling regardless of server as asterisk still will not be able to record call in HD but some other switches like FreeSWITCH can do it or you need to write your own app like it. It's not difficult at all to perform what you're referring to..If you have the hardware... A simple way is to have a port on your main network switch/router that will firehose the traffic the device interacts with In case someone reading this doesn't know, I'm talking about having a port that just makes a copy of EVERY PACKET that the device sees and sends those copies out over the port that you've set up for the purpose..It just GUSHES data over that port...like a firehose just gushes out all the water it possibly can... LOL Anyway, once your data is being mirrored over that firehose, send it to a dedicated recording server...all it has to do is find the signaling packets for each call and then just dump the payload from the RTP. It'll come out exactly as it was transported within RTP...in the codec the call set up I may be wrong, but I'm fairly sure that Asterisk can write a filetype for almost any of it's codecs...I know it can READ audio files that are encoded in GSM, uLaw (ul), aLaw (al), G726 and G729 formats (.g729, g.726)...etc... If the DECoding portion is there, there's almost GOT to be the enCOding functionality... -- _ -- 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] Call files error
However the two calls are placed, the CDRs and the callerids are set correctly, we can't hear each other. As I saw in the logs, the problem is that the calls are placed in the same context, and not being connected ( like one call, but with the variable EXTEN changed ). I'm really confused about doing this, so can you please advise? Thanks, Tamas Tamas, Try appending /n to both of your Local channel definitions... literally a forward slash and a lowercase n...not newline :D -- _ -- 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] Set variable on Call Answer
the M option in your Dial command will execute a macro upon connection, there's also an option to perform a Gosub... http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial ;-) *keeps his mailing-list police badge in it's box in his office* (that wasn't directed at you Dan...there was a little flamewar that I stirred up the other day..that was my troll bit for the day) Check out that link, or run core show application dial from the Asterisk console..look at the options list and find the Macro reference and the Gosub reference...they should light a candle for ya :D On Tue, Feb 8, 2011 at 8:36 AM, Dan Dan dani.mani...@gmail.com wrote: Hi All, First post here. I am dialing out via call file to remote number, when call is connected a local number is dialed. And on success both calls get bridged and works fine. This is a parallel auto dialout application. I want to set a variable as soon as the local number answers the call, so that system won't try to dialout that local number again and stops further dialing. What should be the best way to deal this situation ? Any help would be appreciated. Thanks -dani -- _ -- 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] Callback through extensions.conf?
On Mon, Feb 7, 2011 at 2:22 AM, Gilles codecompl...@free.fr wrote:
Lowering it to 5 seconds makes no difference. I also tried adding a
Hangup before Wait but then the script ends before Wait.
That's just CRAZY mate! I'm thinking it has EVERYTHING to do with your
DAHDI/Zap setup... Barring something in your configuration that I don't know
about, there's no reason that the system should just hang up the call during
the Wait() command...
Could it be that it's just not possible to reuse a channel to dial out
after it's been used to receive a call, even though it was just for a
ring?
Well, first of all, the channel (in the example dialplan and logs you posted
earlier) wouldn't even be dialing a call, it would just be responsible for
the generation of the callfile that would then cause Asterisk to spawn a
call via whatever Channel you specified
I just had a thought thoughAre you, perhaps, hanging your mobile (or
whatever) phone up after dialing into the system to trigger that context?
The reason I ask is that would make this suddenly seem more clear
Basically, try this modified version of the dialplan code:
[from_fxo]
exten = s,1,Wait(2)
exten = s,n,Set(SOURCE_CIDNUMBER=${
CALLERID(num)})
exten = s,n,Set(SOURCE_CIDNAME=${CALLERID(name)})
exten = s,n,NoOp(Call from ${SOURCE_CIDNAME} - ${SOURCE_CIDNUMBER})
exten = s,n,GotoIf($[${SOURCE_CIDNUMBER} = ${GSM}]?goodcid:badcid)
exten = s,n(goodcid),NoOp(CID OK)
;how to reliably detect that line is now quiet?
exten = s,n,Wait(10)
; Note From Sherwood McGowan
; By Changing the exten = s to exten = h in the section below, we
guarantee that Asterisk will execute the code IF THE CALL IS ENDED (like in
the examples given on the mailing list)
; Good Luck!
exten = h,1,NoOp(Before cp)
exten = h,n,system(cp /var/spool/asterisk/skelett.call
/var/tmp/skelett.call)
exten = h,n,NoOp(Before echo)
exten = h,n,system(echo Channel: ZAP/1/${IPPI} /var/tmp/skelett.call)
exten = h,n,NoOp(Before mv)
exten = h,n,system(mv /var/tmp/skelett.call /var/spool/asterisk/outgoing/)
Cheers!
--
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