Re: [asterisk-users] setting variable for a DID number
On Thu, 19 Aug 2010, Tino wrote:
But when i call my DID number following dialplans are being executed.
What i need is to set a variable with one value for one DID number and
set the same variable with another value for another DID number. Also
any contexts should be able to use this variable.
On Thu, Aug 19, 2010 at 11:01 PM, Steve Edwards
asterisk@sedwards.com wrote:
exten = _x.,n, set(FOO=XXX)
exten = _x.,n, execif($[${ANI} =
551212],set,FOO=YYY})
On Thu, 19 Aug 2010, Sherwood McGowan wrote:
Oh boy...Ok, first, let's get into the root issue here...First, when a
variable is set during a call, unless it's defined as a GLOBAL variable,
it is only accessible to THE CHANNEL THAT EXECUTED THE SET COMMAND THAT
CAUSED THE VARIABLE TO EXIST.
[snip]
On Thu, 19 Aug 2010, Sherwood McGowan wrote (in an unrelated post):
I'll leave the Surely you thought of checking THIS discussion for when
I'm a little less likely to spill Jameson and/or Guiness on me lappy
I think you got more down your gullet than in your lap :)
The OP said any contexts not any channels.
Sherwood Mother-F'ing McGowan
Because I'm the Mickand I'm awesome
P.S. a Sixpack of your choosing to the first person who can correctly
identify the person or character the last line of that signature was
parodying
Depending on what you find entertaining, either Miz (WWE) or Barney
(HIMYM).
Boddington's will be most acceptable. Unfortunately, on this side of the
pond they only sell it in a 4pack.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000--
_
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Re: [asterisk-users] Executing system commands through Manager API
On Thu, 2010-08-19 at 16:56 -0500, Carlos Chavez wrote: I am making a web interface so users can manage their voicemail. The only problem I have is that since the Web server and Asterisk run as different users I need to run some commands through Asterisk so I can manipulate the voicemail files. I know that from the CLI I can user the ! commando to run any external shell command but when I try to do it from the Manager API using Command I cannot get it to work. Since the web server cannot erase or modify files I need to go through Asterisk to execute rm or mv. Is there an easier way to do this (without changing the user for Apache)? Is it possible to use the ! command from the Manager? -- _ -- 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 Hi We had a similar, though not exactly the same issue and got round it by putting the apache user into the asterisk group and made all the folders and files in /var/spool/asterisk/voicemail group writeable. I think you'd need to change the umask on it too. This way the apache user can delete the files. However, please not that there is a trade off between security and convenience here although it isn't horrendously insecure. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling Line Identity - any ideas
Paddy, I believe I have a solution, let me sober a bit ;) and rum it through (typo not intended but funny) my test server to doublecheck Sent from my iPhone On Aug 20, 2010, at 12:20 AM, Paddy Grice pa...@wizaner.com wrote: Hi Sherwood I actually do want dynamic CLID as I tried to make clearer I don't know if this makes it any clearer - An internal call from Ext123 should send 123 as the CLID to SIP/ Ext400 but should send 442071110123 to SIP/TheWorld but an external call from 44123455667788 should send the received CLID 44123455667788 to both. So over the provider connection the CLID will be different for different calls. Setting the main office number in sip.conf is fine as a default but as the code/dialplan needs to set cli for some calls I actually set CLID for all calls. This setting and onward transmission by provider works fine. So what I am trying to do is call 2 different sip endpoints AT THE SAME TIME presenting different AND VARIABLE CLIs. If Nasir's trick is not recommended what is the best way to achieve this. As a newbie to Asterisk advise and best practice gained from user experience is always welcome. Paddy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sherwood McGowan Sent: 20 August 2010 04:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calling Line Identity - any ideas On Thu, Aug 19, 2010 at 7:46 PM, Nasir Iqbal na...@ictinnovations.com wrote: Hi, there's still no conceivable reason What can be? except performance! (as asterisk has to create one additional leg and bridge it) Which is very conceivable to those who are dealing with high load traffic. And what will be the option, if other outgoing call requires different custom CLI while using the same trunk? Regards -- Nasir Iqbal ICT Innovations http://www.ictinnovations.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 First, the reason is, why use a BAD IDEA when there's perfectly good solutions in front of the user There was no mention on this ONE call going outbound over the trunk needing a different CID...the request was as follows: Client needs to call an INTERNAL extension, where the INTERNAL CallerID will be used, and at the SAME TIME, a call to an EXTERNAL number (which would necessitate USING THEIR PROVIDER TRUNK), using the EXTERNAL CallerID Now, p-lease tell me how just configuring the damned trunk's outbound CID is NOT more sensible, efficient, and just friggin' COMMON SENSE TO START WITH...over using a Local channel call, which would require slightly more typing, and using something that I've almost NEVER found a good reason to use, and if you'd care to search the damn archives, you'll see that I was pushing upwards of 5k CONCURRENT CALLS back in 2005, WITH 1.4 Trunk and the RealTime addiiton (which was experimental)... For the love of whatever you find holy and good and true...don't come at me like that...I'm really not in the mood anymore...I put 3-4 solid years of helpjng newbies figure out why shit didn't work, reporting REAL bugs and issues to thew developers and even assisting with some of the fixesI feel entitled (yes, I know that's an asshole thing to say) to a little common respect Now...anyone for a pint? I'm off to vent some frustration with people who jump on the WRONG bandwagon and try to take over Sherwood Mother-F'in' McGowanb... Telecommunications and Tattooing You konw anyone else who combines those two professions? I'd like to buy that guy a drink! -- _ -- 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] setting variable for a DID number
A boddingtons...
Anyway, let me point out that the CONTEXT has nothing to do with
'access' to a variable...if a call (channel) causes a variable to be
assigned a value, then that calland possibly it's 'children' if
inheritance is set up. It doesntmatter what context the call ends up
being routed to, it will ALWAYs have acces to that variable unless it
wascreated as a LOCAL variable in a macro
Now, I knew the OP said context, but context does not matter, asterisk
looks to see which CHANNEL has access to a variable's instanceI'll
behappy tofurther expound upon thiswhen I get back to mylaptop
Sent from my iPhone
On Aug 20, 2010, at 2:09 AM, Steve Edwards asterisk@sedwards.com
wrote:
On Thu, 19 Aug 2010, Tino wrote:
But when i call my DID number following dialplans are being
executed. What i need is to set a variable with one value for one
DID number and set the same variable with another value for
another DID number. Also any contexts should be able to use this
variable.
On Thu, Aug 19, 2010 at 11:01 PM, Steve Edwards asterisk@sedwards.com
wrote:
exten = _x.,n, set(FOO=XXX)
exten = _x.,n, execif($[${ANI} =
551212],set,FOO=YYY})
On Thu, 19 Aug 2010, Sherwood McGowan wrote:
Oh boy...Ok, first, let's get into the root issue here...First,
when a variable is set during a call, unless it's defined as a
GLOBAL variable, it is only accessible to THE CHANNEL THAT EXECUTED
THE SET COMMAND THAT CAUSED THE VARIABLE TO EXIST.
[snip]
On Thu, 19 Aug 2010, Sherwood McGowan wrote (in an unrelated post):
I'll leave the Surely you thought of checking THIS discussion for
when I'm a little less likely to spill Jameson and/or Guiness on me
lappy
I think you got more down your gullet than in your lap :)
The OP said any contexts not any channels.
Sherwood Mother-F'ing McGowan
Because I'm the Mickand I'm awesome
P.S. a Sixpack of your choosing to the first person who can
correctly identify the person or character the last line of that
signature was parodying
Depending on what you find entertaining, either Miz (WWE) or Barney
(HIMYM).
Boddington's will be most acceptable. Unfortunately, on this side of
the pond they only sell it in a 4pack.
--
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 --
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--
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Re: [asterisk-users] setting variable for a DID number
On Thu, 19 Aug 2010, Tino wrote: But when i call my DID number following dialplans are being executed. What i need is to set a variable with one value for one DID number and set the same variable with another value for another DID number. Also any contexts should be able to use this variable. On Fri, 20 Aug 2010, Sherwood McGowan wrote: Anyway, let me point out that the CONTEXT has nothing to do with 'access' to a variable...if a call (channel) causes a variable to be assigned a value, then that calland possibly it's 'children' if inheritance is set up. It doesntmatter what context the call ends up being routed to, it will ALWAYs have acces to that variable unless it wascreated as a LOCAL variable in a macro Now, I knew the OP said context, but context does not matter, asterisk looks to see which CHANNEL has access to a variable's instanceI'll behappy tofurther expound upon thiswhen I get back to mylaptop Without clarification from the OP as to whether he meant context (indicating he doesn't understand the scope of a channel variable) or meant channel (indicating he may have had as much to drink as yourself) further discussion is pointless :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting variable for a DID number
Hey Steve, I'm not drunk. I was attempting to keep some levity in my posts because I was pretty frickin irate at the time of several posts that night. Or, maybe I AM drunk, and I can still remember all sorts of nifty Asterisk dialplan and related administration stuff while plastered Or, it's the third option, which involves heavy bouts with insomnia over the last monthit's 5 AM here in StLouis MOSlainte!:D Sent from my iPhone On Aug 20, 2010, at 4:33 AM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 19 Aug 2010, Tino wrote: But when i call my DID number following dialplans are being executed. What i need is to set a variable with one value for one DID number and set the same variable with another value for another DID number. Also any contexts should be able to use this variable. On Fri, 20 Aug 2010, Sherwood McGowan wrote: Anyway, let me point out that the CONTEXT has nothing to do with 'access' to a variable...if a call (channel) causes a variable to be assigned a value, then that calland possibly it's 'children' if inheritance is set up. It doesntmatter what context the call ends up being routed to, it will ALWAYs have acces to that variable unless it wascreated as a LOCAL variable in a macro Now, I knew the OP said context, but context does not matter, asterisk looks to see which CHANNEL has access to a variable's instanceI'll behappy tofurther expound upon thiswhen I get back to mylaptop Without clarification from the OP as to whether he meant context (indicating he doesn't understand the scope of a channel variable) or meant channel (indicating he may have had as much to drink as yourself) further discussion is pointless :) -- 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] Click2call from an OpenOffice document
Make a html link this way: a href= http://widget.doddlephone.com/embed/webphone.jsp?sipserver=SERVERusername=USERpassword=PASSWORDcallto=PHONETOCALLauto=yes Tel: +1 234 567 890 /a /b Sergio On Fri, Jul 9, 2010 at 5:29 AM, Olivier oza_4...@yahoo.fr wrote: Hi, What would you suggest to get click2call from an OpenOffice document ? For instance, in OOo Writer, there is a block : M. John Doe Tel: +1 234 567 890 email: j...@example.com Looking at this block, the line +1 234 567 890 is underlined. When clicking on this, a contextual menu pops up allowing you to make a call. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Codec choice
Hi, Thanks. Actually can it be done on whole kit basis rather than for an extension or peer. Like if there are lot of inbound sip interconnects on a kit , how can we send first 50% simultaneous calls to dahdi with codec A and after that with codec B. Thanks, 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
[asterisk-users] Push to talk over cellular
Hi, i'm trying to get PoC on Nokia Phones to work with asterisk. I think the store-and-forward part could easy be done in the dialplan, but i can't even get the handset to register with asterisk (authentication failed). I'd try'd to find the difference between pure sip and PoC-SIP, but didn't suceed. Has someone an idea how to get this to work? Regards, Jay -- _ -- 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 + openBTS
On 19 Aug 2010, at 20:59, Randy R wrote: On Thu, Aug 19, 2010 at 12:37 PM, Alan Lord (News) alansli...@gmail.com wrote: On 19/08/10 18:20, equis software wrote: I want to know about asterisk and openBTS This island runs it's GSM network on OpenBTS: http://www.niueisland.com/ This was the place he presented about. Read the blog here: http://openbts.sourceforge.net/NiuePilot/ and more about the installation here: http://vuc.me/2010/island-telephony-adventure/ I was part of the team that went to Niue to install OpenBTS, I'm happy to answer questions if you have them, although I'm not the radio guy - asterisk is more my thing :-) Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- 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] Click2call from an OpenOffice document
2010/8/20 Doddle WebPhone doddleph...@gmail.com Make a html link this way: a href= http://widget.doddlephone.com/embed/webphone.jsp?sipserver=SERVERusername=USERpassword=PASSWORDcallto=PHONETOCALLauto=yes Tel: +1 234 567 890 /a /b Sergio Hi, Yes, adding this kind of link should do it but I'm looking for a solution which automatically insert whatever is needed to launch a call. This feature is called Smartags in MSOffice (see http://en.wikipedia.org/wiki/Smart_tag_(Microsoft)) is mostly used for Spellchecking. On Fri, Jul 9, 2010 at 5:29 AM, Olivier oza_4...@yahoo.fr wrote: Hi, What would you suggest to get click2call from an OpenOffice document ? For instance, in OOo Writer, there is a block : M. John Doe Tel: +1 234 567 890 email: j...@example.com Looking at this block, the line +1 234 567 890 is underlined. When clicking on this, a contextual menu pops up allowing you to make a call. Regards -- _ -- 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] asterisk + openBTS
On Fri, Aug 20, 2010 at 10:41 AM, Tim Panton t...@westhawk.co.uk wrote: On 19 Aug 2010, at 20:59, Randy R wrote: On Thu, Aug 19, 2010 at 12:37 PM, Alan Lord (News) alansli...@gmail.com wrote: On 19/08/10 18:20, equis software wrote: I want to know about asterisk and openBTS This island runs it's GSM network on OpenBTS: http://www.niueisland.com/ This was the place he presented about. Read the blog here: http://openbts.sourceforge.net/NiuePilot/ and more about the installation here: http://vuc.me/2010/island-telephony-adventure/ I was part of the team that went to Niue to install OpenBTS, I'm happy to answer questions if you have them, although I'm not the radio guy - asterisk is more my thing :-) Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk In all reality, Asterisk could be substituted with any other platform. All the magic happens in the USRP, OpenBTS, and the cellular phones. Asterisk is merely handling the routing and voice, same as it ever was. It is just the top of the stack. I have two USRPs and a handful of daughter boards, and yes I have two flex 800s that have been physically altered so they can also be flex 1800s with a simple command line. These are the boards you want for GSM (Cellular). There is also a project to be able to listen into phone calls (thanks to the French making encryption so weak) besides a ton of other applications that can be dreamed up. You can do passive radar, track people that have cell phones powered on, RFID (Free tolls anyone?), WiFi, heck, you can even kill people with certain types of pacemakers. While OpenBTS is cool and is on topic with Asterisk, read up on GNURadio and all the projects and applications you can come up with. It is really cool technology. Start here http://gnuradio.org/redmine/projects/show/gnuradio but you can easily find things like this http://tech.mit.edu/V128/N30/subway/Defcon_Presentation.pdf or come up with your own with a bit of imagination and skillz. 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
Re: [asterisk-users] billsec exceeds duration on some calls
On Wednesday 11 Aug 2010, Tilghman Lesher wrote: On Wednesday 11 August 2010 03:59:28 A J Stiles wrote: I'm having a problem with Asterisk 1.6.2.9 with the MySQL cdr addon. With some calls, the value in the `billsec` field in the CDR is exceeding the value in the `duration` field. I'd love to know what circumstance caused that. I agree that this should not occur. I've done some more digging about. I was getting calls in the monitor folder where the outgoing and incoming halves were different lengths; so I temporarily disabled removing them after combining them into a single file, and let them build up for a few days. There doesn't seem to be any correlation between this phenomenon and billsec being duration, though. Can anyone else with a similar setup try running a query such as SELECT COUNT(*) FROM cdr WHERE calldate LIKE 2010-08-20% AND billsecduration ; and seeing if they have any calls like this? -- AJS -- _ -- 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] Calling Line Identity - any ideas
With all honor and respect you deserve, Do I need your permission to express my point of view on community forum ? also it would be quiet helpful for us if you understand well the requirement of post Regards On Fri, Aug 20, 2010 at 1:34 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: Paddy, I believe I have a solution, let me sober a bit ;) and rum it through (typo not intended but funny) my test server to doublecheck Sent from my iPhone On Aug 20, 2010, at 12:20 AM, Paddy Grice pa...@wizaner.com wrote: Hi Sherwood I actually do want dynamic CLID as I tried to make clearer I don't know if this makes it any clearer - An internal call from Ext123 should send 123 as the CLID to SIP/ Ext400 but should send 442071110123 to SIP/TheWorld but an external call from 44123455667788 should send the received CLID 44123455667788 to both. So over the provider connection the CLID will be different for different calls. Setting the main office number in sip.conf is fine as a default but as the code/dialplan needs to set cli for some calls I actually set CLID for all calls. This setting and onward transmission by provider works fine. So what I am trying to do is call 2 different sip endpoints AT THE SAME TIME presenting different AND VARIABLE CLIs. If Nasir's trick is not recommended what is the best way to achieve this. As a newbie to Asterisk advise and best practice gained from user experience is always welcome. Paddy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sherwood McGowan Sent: 20 August 2010 04:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calling Line Identity - any ideas On Thu, Aug 19, 2010 at 7:46 PM, Nasir Iqbal na...@ictinnovations.com wrote: Hi, there's still no conceivable reason What can be? except performance! (as asterisk has to create one additional leg and bridge it) Which is very conceivable to those who are dealing with high load traffic. And what will be the option, if other outgoing call requires different custom CLI while using the same trunk? Regards -- Nasir Iqbal ICT Innovations http://www.ictinnovations.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 First, the reason is, why use a BAD IDEA when there's perfectly good solutions in front of the user There was no mention on this ONE call going outbound over the trunk needing a different CID...the request was as follows: Client needs to call an INTERNAL extension, where the INTERNAL CallerID will be used, and at the SAME TIME, a call to an EXTERNAL number (which would necessitate USING THEIR PROVIDER TRUNK), using the EXTERNAL CallerID Now, p-lease tell me how just configuring the damned trunk's outbound CID is NOT more sensible, efficient, and just friggin' COMMON SENSE TO START WITH...over using a Local channel call, which would require slightly more typing, and using something that I've almost NEVER found a good reason to use, and if you'd care to search the damn archives, you'll see that I was pushing upwards of 5k CONCURRENT CALLS back in 2005, WITH 1.4 Trunk and the RealTime addiiton (which was experimental)... For the love of whatever you find holy and good and true...don't come at me like that...I'm really not in the mood anymore...I put 3-4 solid years of helpjng newbies figure out why shit didn't work, reporting REAL bugs and issues to thew developers and even assisting with some of the fixesI feel entitled (yes, I know that's an asshole thing to say) to a little common respect Now...anyone for a pint? I'm off to vent some frustration with people who jump on the WRONG bandwagon and try to take over Sherwood Mother-F'in' McGowanb... Telecommunications and Tattooing You konw anyone else who combines those two professions? I'd like to buy that guy a drink! -- _ -- 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
Re: [asterisk-users] asterisk + openBTS
Hi Tim, I'm not a radio guy too! I saw your name on the test in Niue. I have a softswitch. Can I replace Asterisk by my softswitch? Thanks On Fri, Aug 20, 2010 at 12:32 PM, Steve Totaro stot...@totarotechnologies.com wrote: On Fri, Aug 20, 2010 at 10:41 AM, Tim Panton t...@westhawk.co.uk wrote: On 19 Aug 2010, at 20:59, Randy R wrote: On Thu, Aug 19, 2010 at 12:37 PM, Alan Lord (News) alansli...@gmail.com wrote: On 19/08/10 18:20, equis software wrote: I want to know about asterisk and openBTS This island runs it's GSM network on OpenBTS: http://www.niueisland.com/ This was the place he presented about. Read the blog here: http://openbts.sourceforge.net/NiuePilot/ and more about the installation here: http://vuc.me/2010/island-telephony-adventure/ I was part of the team that went to Niue to install OpenBTS, I'm happy to answer questions if you have them, although I'm not the radio guy - asterisk is more my thing :-) Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk In all reality, Asterisk could be substituted with any other platform. All the magic happens in the USRP, OpenBTS, and the cellular phones. Asterisk is merely handling the routing and voice, same as it ever was. It is just the top of the stack. I have two USRPs and a handful of daughter boards, and yes I have two flex 800s that have been physically altered so they can also be flex 1800s with a simple command line. These are the boards you want for GSM (Cellular). There is also a project to be able to listen into phone calls (thanks to the French making encryption so weak) besides a ton of other applications that can be dreamed up. You can do passive radar, track people that have cell phones powered on, RFID (Free tolls anyone?), WiFi, heck, you can even kill people with certain types of pacemakers. While OpenBTS is cool and is on topic with Asterisk, read up on GNURadio and all the projects and applications you can come up with. It is really cool technology. Start here http://gnuradio.org/redmine/projects/show/gnuradio but you can easily find things like this http://tech.mit.edu/V128/N30/subway/Defcon_Presentation.pdf or come up with your own with a bit of imagination and skillz. 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
Re: [asterisk-users] Click2call from an OpenOffice document
On Friday, August 20, 2010 10:35:10 am Olivier wrote: Yes, adding this kind of link should do it but I'm looking for a solution which automatically insert whatever is needed to launch a call. wouldn't it be difficult to know exactly which applications are available on the system which has the document open? the solution might be different for every reader of that document. the previously proposed web link-based solution would provide you with the greatest reach. perhaps we aren't exactly sure what you are trying to accomplish. what is your end goal? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] WaitExten() always times out
Thanks for all the help, but I still can't find what's wrong. I enabled console = notice,warning,error,debug,dtmf like Miguel suggested. The output is attached. I noticed that the rtp.c session never starts, which as I understand is what catches the dtmf tone, but I could not find how to start it :s. The Answer() and waitExten(5,m) didn't fix my problem. I hope someone can help me see the problem after looking at the attached console output. On Thu, Aug 19, 2010 at 2:46 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Miguel Molina *Subject:* Re: [asterisk-users] WaitExten() always times out snip Til gave you the answer; When you call out the other end controls timing. Put a waitexten(5,m) in front of background(welcome) and see if that helps -- _ -- 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 -- Executing [...@default:1] Answer(SIP/xx.xx.xxx.xx-0026, ) in new stack [Aug 20 16:50:04] DEBUG[5319]: chan_sip.c:6197 sip_answer: SIP answering channel: SIP/xx.xx.xxx.xx-0026 [Aug 20 16:50:04] DEBUG[5319]: chan_sip.c:10426 transmit_response_with_sdp: Setting framing from config on incoming call [Aug 20 16:50:04] DEBUG[5319]: chan_sip.c:10115 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: True Text flag: True [Aug 20 16:50:04] DEBUG[5319]: chan_sip.c:10116 add_sdp: ** Our prefcodec: 0x0 (nothing) [Aug 20 16:50:04] DEBUG[5319]: chan_sip.c:10227 add_sdp: -- Done with adding codecs to SDP [Aug 20 16:50:04] DEBUG[5319]: channel.c:3096 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan-timingfd=29) [Aug 20 16:50:04] DEBUG[5319]: chan_sip.c:10363 add_sdp: Done building SDP. Settling with this capability: 0xc (ulaw|alaw) [Aug 20 16:50:04] DEBUG[5319]: chan_sip.c:3557 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for xx.xx.xxx.xx:5060 [Aug 20 16:50:04] DEBUG[1188]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for SIP - xx.xx.xxx.xx [Aug 20 16:50:04] DEBUG[1188]: chan_sip.c:23000 sip_devicestate: Checking device state for peer xx.xx.xxx.xx [Aug 20 16:50:04] DEBUG[1188]: devicestate.c:460 do_state_change: Changing state for SIP/xx.xx.xxx.xx - state 2 (In use) [Aug 20 16:50:04] DEBUG[1188]: devicestate.c:440 devstate_event: device 'SIP/xx.xx.xxx.xx' state '2' [Aug 20 16:50:04] DEBUG[1188]: devicestate.c:342 _ast_device_state: No provider found, checking channel drivers for SIP - xx.xx.xxx.xx [Aug 20 16:50:04] DEBUG[1188]: chan_sip.c:23000 sip_devicestate: Checking device state for peer xx.xx.xxx.xx [Aug 20 16:50:04] DEBUG[1188]: devicestate.c:460 do_state_change: Changing state for SIP/xx.xx.xxx.xx - state 2 (In use) [Aug 20 16:50:04] DEBUG[1188]: devicestate.c:440 devstate_event: device 'SIP/xx.xx.xxx.xx' state '2' [Aug 20 16:50:04] DEBUG[1251]: app_queue.c:1084 handle_statechange: Device 'SIP/xx.xx.xxx.xx' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 20 16:50:04] DEBUG[1251]: app_queue.c:1084 handle_statechange: Device 'SIP/xx.xx.xxx.xx' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Aug 20 16:50:04] DEBUG[5319]: channel.c:1882 __ast_answer: Didn't receive a media frame from SIP/xx.xx.xxx.xx-0026 within 500 ms of answering. Continuing anyway [Aug 20 16:50:04] DEBUG[5319]: pbx.c:3692 pbx_extension_helper: Launching 'WaitExten' -- Executing [...@default:2] WaitExten(SIP/xx.xx.xxx.xx-0026, 10,m) in new stack -- Started music on hold, class 'default', on SIP/xx.xx.xxx.xx-0026 [Aug 20 16:50:04] DEBUG[5319]: channel.c:2426 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 20 16:50:04] DEBUG[5319]: channel.c:3727 set_format: Set channel SIP/xx.xx.xxx.xx-0026 to write format slin [Aug 20 16:50:04] DEBUG[5319]: res_musiconhold.c:303 ast_moh_files_next: SIP/xx.xx.xxx.xx-0026 Opened file 1 '/var/lib/asterisk/moh/manolo_camp-morning_coffee' [Aug 20 16:50:04] DEBUG[5319]: rtp.c:3832 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 20 16:50:04] DEBUG[5319]: rtp.c:3858 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 20 16:50:05] DEBUG[1232]: chan_sip.c:3758 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #6725)) [Aug 20 16:50:05] DEBUG[1232]: chan_sip.c:3557 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for xx.xx.xxx.xx:5060 [Aug 20 16:50:06] DEBUG[1232]: chan_sip.c:3758
Re: [asterisk-users] Codec choice
1. Set up a Global Variable that will store that kit's current number of calls 2. Check that variable when a call starts (but before you dial out) 3. If the number of calls is 49 (since the current call will make 50), use codec A via the CHANNEL() function, otherwise use codec B using the same function. 4. Increment the variable 5. place call 6., upon hangup, decrement the variable Cheers On Fri, Aug 20, 2010 at 9:06 AM, Deepika Nijhawan deepika.nijha...@oxygen8.com wrote: Hi, Thanks. Actually can it be done on whole kit basis rather than for an extension or peer. Like if there are lot of inbound sip interconnects on a kit , how can we send first 50% simultaneous calls to dahdi with codec A and after that with codec B. Thanks, 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 -- _ -- 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] Calling Line Identity - any ideas
Nasir Iqbal na...@ictinnovations.com wrote: With all honor and respect you deserve, Do I need your permission to express my point of view on community forum ? also it would be quiet helpful for us if you understand well the requirement of post *snip* Nasir, You don't need my permission to post on a public forum...However, neither do I, and I took issue with what you said, and found that your comment about those who are dealing with high load traffic offensive, since it made the assumption that I was just some new guy who deals with hobby/small Asterisk systems and doesn't know what he's talking aboutTherefore, I made it abundantly clear that I wasn't, and that I definitely took issue with that statement. However, I will say that yes, I did mis-take something the OP said... Paddy: Now, here's idea I came up with (haven't tested yet, too busy writing a system for an international interpretation company's telecom needs) First of all, you should have a separate context for outbound calls made by internal extensions... so, in THAT context have code to set the CID to what you wish (you can do logic control and if you're feeling spiffy you can even lookup what CLID to use based on the extension making the call). Second, calls that are being passed from the outside world onto should pass through a different context, performing pretty much the same function... Third, both of THOSE contexts should then pass to a third context that performs the dialout using the multiple targets... Let me know if that works...I know I can make this do what you want, but I'm not trying to do all the work, just point you in a direction, since I get paid to actually do the work ;-) Cheers all, and remember, some of us have been doing this a while, and get grumpy... ;-) there's still no conceivable reason What can be? except performance! (as asterisk has to create one additional leg and bridge it) Which is very conceivable to those who are dealing with high load traffic. And what will be the option, if other outgoing call requires different custom CLI while using the same trunk? 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 First, the reason is, why use a BAD IDEA when there's perfectly good solutions in front of the user There was no mention on this ONE call going outbound over the trunk needing a different CID...the request was as follows: Client needs to call an INTERNAL extension, where the INTERNAL CallerID will be used, and at the SAME TIME, a call to an EXTERNAL number (which would necessitate USING THEIR PROVIDER TRUNK), using the EXTERNAL CallerID Now, p-lease tell me how just configuring the damned trunk's outbound CID is NOT more sensible, efficient, and just friggin' COMMON SENSE TO START WITH...over using a Local channel call, which would require slightly more typing, and using something that I've almost NEVER found a good reason to use, and if you'd care to search the damn archives, you'll see that I was pushing upwards of 5k CONCURRENT CALLS back in 2005, WITH 1.4 Trunk and the RealTime addiiton (which was experimental)... For the love of whatever you find holy and good and true...don't come at me like that...I'm really not in the mood anymore...I put 3-4 solid years of helpjng newbies figure out why shit didn't work, reporting REAL bugs and issues to thew developers and even assisting with some of the fixesI feel entitled (yes, I know that's an asshole thing to say) to a little common respect Now...anyone for a pint? I'm off to vent some frustration with people who jump on the WRONG bandwagon and try to take over Sherwood Mother-F'in' McGowanb... Telecommunications and Tattooing You konw anyone else who combines those two professions? I'd like to buy that guy a drink! -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ --
Re: [asterisk-users] Codec choice
On Fri, 20 Aug 2010, Sherwood McGowan wrote: 1. Set up a Global Variable that will store that kit's current number of calls 2. Check that variable when a call starts (but before you dial out) 3. If the number of calls is 49 (since the current call will make 50), use codec A via the CHANNEL() function, otherwise use codec B using the same function. 4. Increment the variable 5. place call 6., upon hangup, decrement the variable Not really paying close attention to what you're trying to do, but... The GROUP() and GROUP_COUNT() functions automagically take care of the increment and decrement cruft in a race condition free sort of way. Both methods still leave a small window of opportunity in comparing the count with the threshold. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec choice
Steve, Good point my man...You drinking yet? LOL...I had forgotten about the GROUP and GROUP_COUNT functions, that is a much better way (in that it already existed and doesn't require me to write more code :] ) Slainte! On Fri, Aug 20, 2010 at 7:37 PM, Steve Edwards asterisk@sedwards.com wrote: On Fri, 20 Aug 2010, Sherwood McGowan wrote: 1. Set up a Global Variable that will store that kit's current number of calls 2. Check that variable when a call starts (but before you dial out) 3. If the number of calls is 49 (since the current call will make 50), use codec A via the CHANNEL() function, otherwise use codec B using the same function. 4. Increment the variable 5. place call 6., upon hangup, decrement the variable Not really paying close attention to what you're trying to do, but... The GROUP() and GROUP_COUNT() functions automagically take care of the increment and decrement cruft in a race condition free sort of way. Both methods still leave a small window of opportunity in comparing the count with the threshold. -- 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] Codec choice
On Fri, 20 Aug 2010, Sherwood McGowan wrote: Good point my man...You drinking yet? Let me check to see if I still have a pulse -- yep! -- 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
