Re: [asterisk-users] Calling Line Identity - any ideas
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 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
Re: [asterisk-users] setting variable for a DID number
On Thu, Aug 19, 2010 at 11:01 PM, Steve Edwards
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.
>
> exten = _x.,n, set(FOO=XXX)
> exten = _x.,n, execif($["${ANI}" =
> "551212"],set,FOO=YYY})
>
> --
> 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
>
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". Of course, you can make
it available to child channels that are spawned by the original
channel by prepending the variable name in the Set() command with one
or more underscoresSee the following Voip-Info.org link for more
information regarding channels and variable
inheritancehttp://www.voip-info.org/wiki/view/Asterisk+variables#InheritanceofChannelVariables
If you really need an example of how this works and why variables are
specific to the call they're related to, I'll gladly write one up when
I return from the pub in the wee hours of the morning...if you're
nice...
I'll give you a great real world hint/example
If I was playing a quick round of Doom at a gaming cafe where all the
terminals were thin clients, and all the code to make the game
"happen" was being executed on the application server, my game would
not suddenly jump back to the first level when you told the
application server you wanted to play Doom as wellNow, envision
the application server as your Asterisk server, and our respective
games as calls coming into that Asterisk server..
Getting a candle? Possibly a torch? Maybemaybea light bulb?
Let me know, I'll gladly elucidate further if need be, but be aware
that it will be after a solid round or five of the Slainte
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
--
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Re: [asterisk-users] sending sms from Asterisk server
Un-top-posting... > Un-top-posting... > > On 08/17/2010 09:00 AM, Tino wrote: > > I would like to send sms to some external phone numbers from my asterisk > server. Is it possible to send sms via softphones like X-Lite ? . Any > tips regarding this will be helpful > On Wed, Aug 18, 2010 at 3:13 AM, Johann Hoehn > wrote: > > I personally use an AGI script for this part, but you could use a > System() call as well. > On 18 August 2010 21:02, Steve Edwards > wrote: > Using system() is almost always a hack -- and not the good kind :) On Thu, 19 Aug 2010, Tiago Geada wrote: > I would rather use .call files. So easy to produce a text file... And so hard to make robust! 1) How do you handle a momentary failure like a DNS lookup or a circuit busy? 2) How do you handle a permenent failure like invalid subscriber number or wrong server? 3) How do you confirm success or failure? -- 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] MySQL Connect problem...
On Thu, Aug 19, 2010 at 6:05 PM, Steve Edwards wrote: >>> On 18 August 2010 08:52, Nasir Iqbal wrote: Avoid to use MySQL dialplan application, instead write an AGI script for this purpose > >> On Wed, Aug 18, 2010 at 3:59 PM, Geraint Lee wrote: > >>> This is what I ended up doing, working fine now. Cheers > > On Thu, 19 Aug 2010, Sherwood McGowan wrote: > >> LOL, I hate to say this but writing an AGI script just adds yet another >> application layer to your total solution. > > Yes, another layer, but a layer where you will have full access to what's > going on -- like what errors are being returned by MySQL and can be debugged > completely external from Asterisk via the shell command line. > > -- > 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 > I'll just leave this be...I'm in an irate mood, and I've been using the MySQL addons since 2005...I don't have problems like this, and you CAN see what MySQL is saying...THERE'S LOGS ON BOTH SYSTEMS.. 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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
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.
exten = _x.,n, set(FOO=XXX)
exten = _x.,n, execif($["${ANI}" =
"551212"],set,FOO=YYY})
--
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] Calling Line Identity - any ideas
On Thu, Aug 19, 2010 at 7:46 PM, Nasir Iqbal 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
Re: [asterisk-users] codec_g729.so not work!
On Thu, Aug 19, 2010 at 10:28 PM, Zhang Shukun wrote: > == Using SIP RTP CoS mark 5 > [Aug 20 18:23:21] NOTICE[14543]: chan_sip.c:8454 process_sdp: No > compatible codecs, not accepting this offer! > > Could you tell me what 's wrong? > http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt Then post your debug log, so we can see what is going on. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codec_g729.so not work!
On Thu, Aug 19, 2010 at 9:44 PM, Tim Nelson wrote: > - "Zhang Shukun" wrote: >> >> == Using SIP RTP CoS mark 5 >> [Aug 20 18:23:21] NOTICE[14543]: chan_sip.c:8454 process_sdp: No >> compatible codecs, not accepting this offer! >> >> Could you tell me what 's wrong? > > Besides the use of an illegal G.729 implementation? > > Are both endpoints G.729 compatible? Most hard phones are G.729 compatible > but most softphones do not have G.729 support. Paid softphones typically have > support for G.729... > > --Tim > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every 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'd have to agree with Tim, it sounds like you're needing to transcode between G.729 and another codec because one of your endpoints is not capable of using G729 (or maybe either the sip configuration or the device's configuration is not done properly) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Executing system commands through Manager API
On Thu, Aug 19, 2010 at 5:56 PM, Steve Edwards wrote: > On Thu, 19 Aug 2010, 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? > > 1) Change the user for Asterisk. > > 2) man sudo > > -- > 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 > Another possibility would be to use the IMAP or ODBC storage option for Voicemail? Just a thought :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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_g729.so not work!
- "Zhang Shukun" wrote: > > == Using SIP RTP CoS mark 5 > [Aug 20 18:23:21] NOTICE[14543]: chan_sip.c:8454 process_sdp: No > compatible codecs, not accepting this offer! > > Could you tell me what 's wrong? Besides the use of an illegal G.729 implementation? Are both endpoints G.729 compatible? Most hard phones are G.729 compatible but most softphones do not have G.729 support. Paid softphones typically have support for G.729... --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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_g729.so not work!
hi, all i want to use g729 codec for set up a call. so i donwloaded the so file from web site: http://asterisk.hosting.lv/#bin and install it properly. *CLI> *CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723 - - - -- - - - - - - - - - - - gsm - - 2 2 2000 2 1 3001 3000 - - 2001 1001 - - 2001 ulaw - 3000 - 1 2000 2 1 3001 3000 - - 2001 1001 - - 2001 alaw - 3000 1 - 2000 2 1 3001 3000 - - 2001 1001 - - 2001 g726aal2 - 3999 1001 1001- 1001 1000 4000 3999 - - 3000 2000 - - 3000 adpcm - 3999 1001 1001 2999 - 1000 4000 3999 - - 3000 2000 - - 3000 slin - 2999 1 1 1999 1 - 3000 2999 - - 2000 1000 - - 2000 lpc10 - 4998 2000 2000 3998 2000 1999 - 4998 - - 3999 2999 - - 3999 g729 - 3999 1001 1001 2999 1001 1000 4000 - - - 3000 2000 - - 3000 speex - - - -- - - - - - - - - - - - ilbc - - - -- - - - - - - - - - - - g726 - 3998 1000 1000 2998 1000 999 3999 3998 - - - 1999 - - 2999 g722 - 3998 1000 1000 2998 1000 999 3999 3998 - - 2999 - - - 1000 siren7 - - - -- - - - - - - - - - - - siren14 - - - -- - - - - - - - - - - - slin16 - 4998 2000 2000 3998 2000 1999 4999 4998 - - 3999 1000 - - - my sip.conf add two account: [123] type=friend username=123 host=dynamic context=95040 dtmfmode=rfc2833 disallow=all allow=g729 insecure=port,invite canreinvite=no nat=yes [321] type=friend username=321 host=dynamic context=95040 dtmfmode=rfc2833 disallow=all allow=g729 insecure=port,invite canreinvite=no my extension is : exten => 321,1,Dial(SIP/321) when i want to set up a call (123 dial 321). but failed. it says: == Using SIP RTP CoS mark 5 [Aug 20 18:23:21] NOTICE[14543]: chan_sip.c:8454 process_sdp: No compatible codecs, not accepting this offer! Could you tell me what 's wrong? -- Thanks & Regards Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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
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
Re: [asterisk-users] MySQL Connect problem...
On 18 August 2010 08:52, Nasir Iqbal wrote: Avoid to use MySQL dialplan application, instead write an AGI script for this purpose On Wed, Aug 18, 2010 at 3:59 PM, Geraint Lee wrote: This is what I ended up doing, working fine now. Cheers On Thu, 19 Aug 2010, Sherwood McGowan wrote: LOL, I hate to say this but writing an AGI script just adds yet another application layer to your total solution. Yes, another layer, but a layer where you will have full access to what's going on -- like what errors are being returned by MySQL and can be debugged completely external from Asterisk via the shell command line. -- 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] Executing system commands through Manager API
On Thu, 19 Aug 2010, 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? 1) Change the user for Asterisk. 2) man sudo -- 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
> On Thu, Aug 19, 2010 at 3:14 AM, Deepika Nijhawan >> >> Does anyone has an idea how to tell asterisk to use codec A for first >> 50 calls and then codec B for rest of the calls. On Thu, 19 Aug 2010, Sherwood McGowan wrote: > the easiest way I can think of is to use a global variable that you > increment each time a new call spawns, and once it's over your threshold > (50 in this case) use the CHANNEL() function to set the audio format to > the codec you want (google voip-info function CHANNEL) Your question is not specific enough. Do you mean the "first 50 calls" or "50 simultaneous calls?" I suspect the latter and the GROUP() and GROUP_COUNT() functions are the way to go. -- 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] Caller ID issue
Sorry for the delay - I lost this message in the middle of a digest. I tried Answer(2000) and was getting an annoying warning: [Aug 15 17:20:11] WARNING[15516]: channel.c:1044 __ast_queue_frame: Exceptionally long voice queue length queuing to DAHDI/1-1 So I changed it back to Wait(2). I'll try shorter wait intervals and see what happens. Cassius > Subject: Re: [asterisk-users] Caller ID issue > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: > > Content-Type: text/plain; charset=ISO-8859-1 > > In most cases wait(.5) will do. I would not recommend using > answer(2000) as that answers the channel, which means you start > getting billed. > > On 8/2/10, Peder wrote: > >> I am using T1's and didn't think the spill would take that long. > > > >> PRI no, E&M yes. > > > > Some PRI take that long too because the telco sends the name in a > followup > > message, not in the initial call setup. > > > > > > -- > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Executing system commands through Manager API
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? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 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
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 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
Re: [asterisk-users] WaitExten() always times out
El 19/08/10 15:07, Kathryn Jones escribió:
Thanks for your reply :)
I added Answer to my dialplan:
exten => xxx,1,Answer()
exten => xxx,n,Background(welcome)
exten => xxx,n,WaitExten(7)
exten => _X,1,AGI(agi.php)
exten => _X,n,PlayBack(vm-tocallnumber)
exten => _X,n,Dial(SIP/voiptrunk/${NUM})
exten => t,1,Noop(*timeout*)
exten => t,n,Playback(pbx-invalid)
exten => t,n,Hangup()
cli output:
-- Executing [...@default:1] Answer("SIP/xx.xx.xx.xx-0004", "") in
new stack
-- Executing [...@default:2]
BackGround("SIP/xx.xx.xx.xx-0004", "welcome") in new stack
-- Playing 'welcome.slin' (language 'en')
-- Executing [...@default:3] WaitExten("SIP/xx.xx.xx.xx-0004",
"7") in new stack
-- Timeout on SIP/xx.xx.xx.xx-0004, going to 't'
-- Executing [...@default:1] NoOp("SIP/xx.xx.xx.xx-0004",
"*timeout*") in new stack
-- Executing [...@default:2] Playback("SIP/xx.xx.xx.xx-0004",
"pbx-invalid") in new stack
-- Playing 'pbx-invalid.gsm' (language
'en')
-- Executing [...@default:3] Hangup("SIP/xx.xx.xx.xx-0004", "")
in new stack
== Spawn extension (default, t, 3) exited non-zero on
'SIP/xx.xx.xx.xx-0004'
[] WARNING[1235]: chan_sip.c:3780 retrans_pkt: Maximum retries
exceeded on transmission 0ef328f40a5fd6ca31a68dae2af75...@xx.xx.xx.xx
for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
[] WARNING[1235]: chan_sip.c:3780 retrans_pkt: Maximum retries
exceeded on transmission 0ef328f40a5fd6ca31a68dae2af75...@xx.xx.xx.xx
for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
I still can't read the DTMF input :(
I also tried adding:
dtmfmode = rfc2833
rfc2833compensate = yes
relaxdmtf = no ; should be no because setting it to yes cause talkoff
to sip.conf and chan_dahdi.conf
and increasing rxgain=20 (I wasn't sure how much was appropriate)
Nothing seems to help.
ANY tips or ideas will be apreciated.
On Thu, Aug 19, 2010 at 1:19 PM, Tilghman Lesher > wrote:
On Wednesday 18 August 2010 16:52:38 Kathryn Jones wrote:
> I must not be receiving them properly, since I can't make it
work. I just
> can't see why :P.
>
> My asterisk version is 1.6.2.6. Like I said before, for outgoing
.call
> files WaitExten works fine, it's on incoming calls that I cannot
receive
> the number I need.
There's your answer. On outgoing calls, the other end signals the
line into
answered state, whereas on incoming calls, you must explicitly
answer the
channel before listening for DTMF.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com &
www.asterisk.org
I suggest you to debug DTMF and core, enabling them in logger.conf:
console => notice,warning,error,debug,dtmf
And issuing a "logger reload" command in asterisk CLI.
A rxgain of 20 is too much for me, leave them in rxgain = 0.0 and
txgain= 0.0. Maybe 20dB gain is high enough to distort the audio signal
and make DTMF detection more difficult.
Look at the DTMF events in your CLI, that way you can debug better. You
can enable core debug if you want issuing the CLI command "core set
debug X", with X on 1 or 2, and setting it off when you want.
If your call is received from the PSTN, asterisk will detect the inband
DTMF tones in the audio signal. The rfc2833 configurations are only for
VoIP endpoints.
Good luck in your debugging,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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Re: [asterisk-users] AstriCon approaches: Innovation Awards, your attendance wanted!
John: August 1 is the deadline??? On Thu, Aug 19, 2010 at 11:58 AM, John Todd wrote: > > Just a reminder: AstriCon is coming up in October in Washington, DC ( > http://www.astricon.net/ > ) and we're looking forward to seeing you there! > > We're getting to the deadline for Innovation Awards for this year. > What's an Innovation Award? The Innovation Award is designed to > recognize developers, customers and partners for outstanding > achievements that are improving business processes, overcoming > technology challenges and enhancing the company's bottom line. Digium > picks five different categories in which certain projects or companies > have excelled in the last year creating amazing things with Asterisk. > The awards are presented at AstriCon. > > If you think you're doing something great with Asterisk, send it in! > It's a great opportunity to be recognized as a leader in Asterisk > development, implementation, and innovation. August 1 is the deadline. > > More details here - > http://www.digium.com/en/company/awards/innovation.php > > Send your Innovation Award proposal to Julie Webb (jw...@digium.com) > for inclusion. > > > AstriCon in general: > I'll take this opportunity to ask everyone again to get your > reservations in for AstriCon this year! We're looking forward to a > really good show, in a city slightly less oven-like than the past > three years. The conference has a fantastic line-up of speakers and > as always, offers the opportunity to talk with people in an informal > setting about their real-world experiences with Asterisk, VoIP, > different hardware, methods of implementation, and make all sorts of > connections that you just can't get without meeting face-to-face. > Washington DC is convenient from Europe, with direct flights to IAD > (Dulles), DCA (Reagan International), and BWI (Baltimore Washington) > airports from most major European and South American cities. > > JT > > > --- > John Todd > email:jt...@digium.com > Digium, Inc. | Asterisk Open Source Community Director > 445 Jan Davis Drive NW - Huntsville AL 35806 - USA > direct: +1-256-428-6083 http://www.digium.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] WaitExten() always times out
Thanks for your reply :)
I added Answer to my dialplan:
exten => xxx,1,Answer()
exten => xxx,n,Background(welcome)
exten => xxx,n,WaitExten(7)
exten => _X,1,AGI(agi.php)
exten => _X,n,PlayBack(vm-tocallnumber)
exten => _X,n,Dial(SIP/voiptrunk/${NUM})
exten => t,1,Noop(*timeout*)
exten => t,n,Playback(pbx-invalid)
exten => t,n,Hangup()
cli output:
-- Executing [...@default:1] Answer("SIP/xx.xx.xx.xx-0004", "") in new
stack
-- Executing [...@default:2] BackGround("SIP/xx.xx.xx.xx-0004",
"welcome") in new stack
-- Playing 'welcome.slin' (language 'en')
-- Executing [...@default:3] WaitExten("SIP/xx.xx.xx.xx-0004", "7")
in new stack
-- Timeout on SIP/xx.xx.xx.xx-0004, going to 't'
-- Executing [...@default:1] NoOp("SIP/xx.xx.xx.xx-0004",
"*timeout*") in new stack
-- Executing [...@default:2] Playback("SIP/xx.xx.xx.xx-0004",
"pbx-invalid") in new stack
-- Playing 'pbx-invalid.gsm' (language 'en')
-- Executing [...@default:3] Hangup("SIP/xx.xx.xx.xx-0004", "") in new
stack
== Spawn extension (default, t, 3) exited non-zero on
'SIP/xx.xx.xx.xx-0004'
[] WARNING[1235]: chan_sip.c:3780 retrans_pkt: Maximum retries exceeded on
transmission 0ef328f40a5fd6ca31a68dae2af75...@xx.xx.xx.xx for seqno 102
(Critical Response) -- See doc/sip-retransmit.txt.
[] WARNING[1235]: chan_sip.c:3780 retrans_pkt: Maximum retries exceeded on
transmission 0ef328f40a5fd6ca31a68dae2af75...@xx.xx.xx.xx for seqno 102
(Critical Response) -- See doc/sip-retransmit.txt.
I still can't read the DTMF input :(
I also tried adding:
dtmfmode = rfc2833
rfc2833compensate = yes
relaxdmtf = no ; should be no because setting it to yes cause talkoff
to sip.conf and chan_dahdi.conf
and increasing rxgain=20 (I wasn't sure how much was appropriate)
Nothing seems to help.
ANY tips or ideas will be apreciated.
On Thu, Aug 19, 2010 at 1:19 PM, Tilghman Lesher wrote:
> On Wednesday 18 August 2010 16:52:38 Kathryn Jones wrote:
> > I must not be receiving them properly, since I can't make it work. I just
> > can't see why :P.
> >
> > My asterisk version is 1.6.2.6. Like I said before, for outgoing .call
> > files WaitExten works fine, it's on incoming calls that I cannot receive
> > the number I need.
>
> There's your answer. On outgoing calls, the other end signals the line
> into
> answered state, whereas on incoming calls, you must explicitly answer the
> channel before listening for DTMF.
>
> --
> Tilghman Lesher
> Digium, Inc. | Senior Software Developer
> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] asterisk + openBTS
On Thu, Aug 19, 2010 at 12:37 PM, Alan Lord (News) 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/ /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 19/08/10 18:20, equis software wrote: > I want to know about asterisk and openBTS > If anybody made any test and experience... 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/ HTH Al -- The Open Learning Centre http://www.theopenlearningcentre.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WaitExten() always times out
On Wednesday 18 August 2010 16:52:38 Kathryn Jones wrote: > I must not be receiving them properly, since I can't make it work. I just > can't see why :P. > > My asterisk version is 1.6.2.6. Like I said before, for outgoing .call > files WaitExten works fine, it's on incoming calls that I cannot receive > the number I need. There's your answer. On outgoing calls, the other end signals the line into answered state, whereas on incoming calls, you must explicitly answer the channel before listening for DTMF. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk + openBTS
On Thu, Aug 19, 2010 at 10:53 AM, equis software wrote: > May be he was David Burguess, another founder is Harvind Samra ... > Do you know about any Equipment working? > The primary piece of equipment consists of the USRP made by Ettus Research along with driver support provided through GNU Radio. No other equipment is publicly supported at this time, though I would expect that to change in the future as there are other capable devices that exist. The upcoming weeks are an exciting time for OpenBTS as David Burgess, Harvind and others setup their test network at Burning Man. http://pagalegba2010.wikispaces.com/PublicInformation Thomas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Call-limit field
If I set a call-limit field on a peer in users.conf.. I am seeing that it seems to affect other peers too? I am running Asterisk 1.4.18 has someone seen this issue. Peer 1 has call-limit=5 Peer 2 has call-limit=20... In the SIP trace I see when Peer2 hits 5, Asterisk sends back a 480 (temp Unavailable (Call limit reached)...msg.. Any ideas would be appreciated Thx -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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
May be he was David Burguess, another founder is Harvind Samra ... Do you know about any Equipment working? On Thu, Aug 19, 2010 at 2:27 PM, Alan Lord (News) wrote: > On 19/08/10 18:20, equis software wrote: > > I want to know about asterisk and openBTS > > If anybody made any test and experience... > > I saw a presentation a few months ago where one of the openBTS project > founders talked about one early system they set up on a very small and > remote Pacific island along with Asterisk. > > If I can remember/find anything more I'll post here. > > Cheers > > Alan > > -- > The Open Learning Centre > http://www.theopenlearningcentre.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] asterisk + openBTS
On 19/08/10 18:20, equis software wrote: > I want to know about asterisk and openBTS > If anybody made any test and experience... I saw a presentation a few months ago where one of the openBTS project founders talked about one early system they set up on a very small and remote Pacific island along with Asterisk. If I can remember/find anything more I'll post here. Cheers Alan -- The Open Learning Centre http://www.theopenlearningcentre.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
[asterisk-users] asterisk + openBTS
I want to know about asterisk and openBTS If anybody made any test and experience... Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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
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.
-
NoOp("SIP/5070-5407", "Received incoming SIP connection from unknown
peer to ") in new stack
-- Executing [@from-sip-external:2] Set("SIP/5070-5407",
"DID=") in new stack
-- Executing [@from-sip-external:3]
Goto("SIP/5070-5407", "s|1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [...@from-sip-external:1] GotoIf("SIP/5070-5407",
"1?checklang:noanonymous") in new stack
-- Goto (from-sip-external,s,2)
-- Executing [...@from-sip-external:2] GotoIf("SIP/5070-5407",
"0?setlanguage:from-trunk||1") in new stack
-- Goto (from-trunk,,1)
-- Executing [@from-trunk:1] Set("SIP/5070-5407",
"__FROM_DID=") in new stack
-- Executing [@from-trunk:2] Gosub("SIP/5070-5407",
"app-blacklist-check|s|1") in new stack
-- Executing [...@app-blacklist-check:1]
LookupBlacklist("SIP/5070-5407", "") in new stack
-- Executing [...@app-blacklist-check:2] GotoIf("SIP/5070-5407",
"0?blacklisted") in new stack
-- Executing [...@app-blacklist-check:3] Set("SIP/5070-5407",
"CALLED_BLACKLIST=1") in new stack
-- Executing [...@app-blacklist-check:4] Return("SIP/5070-5407", "")
in new stack
-- Executing [@from-trunk:3] ExecIf("SIP/5070-5407", "0
|Set|CALLERID(name)=Anonymous") in new stack
-- Executing [@from-trunk:4] Set("SIP/5070-5407",
"__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [@from-trunk:5]
SetCallerPres("SIP/5070-5407", "allowed_not_screened")
P.S : used in place of actual DID number
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Re: [asterisk-users] Recording the conversation with MixMonitor() ends when the call is transfered
Hi,
Never tried it, but you can take a look to the AUDIOHOOK_INHERIT
function that allows MixMonitor to continue the recording in the same
file after the transfer.
http://www.voip-info.org/wiki/view/Asterisk+func+AUDIOHOOK_INHERIT
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
El 19/08/10 10:49, Jonas Kellens escribió:
Hello list,
A calls B, B transfers to C, A speaks with C.
Anyone knows how I can record a conversation where the call is
transfered ?!
The recording of the call (which begins when B answers) stops when A
and C are connected together.
Can I keep the recording going ?!
Kind regards,
Jonas.
On 08/12/2010 11:01 AM, Jonas Kellens wrote:
Hello.
I notice that when a call that is recorded with MixMonitor is
transfered to another co-worker, the recording ends.
exten => 409,n,Macro(SDstartrecording,external,${DID})
the incoming call then goes to a queue...
[macro-startrecording]
; ARG1 = incoming DID or CALLERID(name)
; ARG2 = outgoing dialnumber
...
exten => s,n,MixMonitor(/var/ftp/${NR}/${recordfile},b,chown -R
${NR}:astDrisk /var/ftp/${NR} && chmod 570 /var/ftp/${NR} && chmod
400 /var/ftp/${NR}/${recordfile})
...
The 'b'-option makes the recording start when the call is answered by
an agent of the queue.
Is it normal that when this agent transfers the call to another
co-worker, the recording stops ?!
The last sound on the recording is that of musiconhold (what the
caller hears when doing an attented transfer). Then it stops...
Kind regards,
Jonas.
--
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[asterisk-users] AstriCon approaches: Innovation Awards, your attendance wanted!
Just a reminder: AstriCon is coming up in October in Washington, DC (http://www.astricon.net/ ) and we're looking forward to seeing you there! We're getting to the deadline for Innovation Awards for this year. What's an Innovation Award? The Innovation Award is designed to recognize developers, customers and partners for outstanding achievements that are improving business processes, overcoming technology challenges and enhancing the company's bottom line. Digium picks five different categories in which certain projects or companies have excelled in the last year creating amazing things with Asterisk. The awards are presented at AstriCon. If you think you're doing something great with Asterisk, send it in! It's a great opportunity to be recognized as a leader in Asterisk development, implementation, and innovation. August 1 is the deadline. More details here - http://www.digium.com/en/company/awards/innovation.php Send your Innovation Award proposal to Julie Webb (jw...@digium.com) for inclusion. AstriCon in general: I'll take this opportunity to ask everyone again to get your reservations in for AstriCon this year! We're looking forward to a really good show, in a city slightly less oven-like than the past three years. The conference has a fantastic line-up of speakers and as always, offers the opportunity to talk with people in an informal setting about their real-world experiences with Asterisk, VoIP, different hardware, methods of implementation, and make all sorts of connections that you just can't get without meeting face-to-face. Washington DC is convenient from Europe, with direct flights to IAD (Dulles), DCA (Reagan International), and BWI (Baltimore Washington) airports from most major European and South American cities. JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.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
[asterisk-users] Loop Detection / SIP
Has anyone found a way to detect a loop condition in the dialplan.?? We had a condition where this filled up 47 PRI channels in an NFAS group connected to our media gateway... and endless loop if you will.. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Recording the conversation with MixMonitor() ends when the call is transfered
Hello list,
A calls B, B transfers to C, A speaks with C.
Anyone knows how I can record a conversation where the call is transfered ?!
The recording of the call (which begins when B answers) stops when A and
C are connected together.
Can I keep the recording going ?!
Kind regards,
Jonas.
On 08/12/2010 11:01 AM, Jonas Kellens wrote:
Hello.
I notice that when a call that is recorded with MixMonitor is
transfered to another co-worker, the recording ends.
exten => 409,n,Macro(SDstartrecording,external,${DID})
the incoming call then goes to a queue...
[macro-startrecording]
; ARG1 = incoming DID or CALLERID(name)
; ARG2 = outgoing dialnumber
...
exten => s,n,MixMonitor(/var/ftp/${NR}/${recordfile},b,chown -R
${NR}:astDrisk /var/ftp/${NR} && chmod 570 /var/ftp/${NR} && chmod 400
/var/ftp/${NR}/${recordfile})
...
The 'b'-option makes the recording start when the call is answered by
an agent of the queue.
Is it normal that when this agent transfers the call to another
co-worker, the recording stops ?!
The last sound on the recording is that of musiconhold (what the
caller hears when doing an attented transfer). Then it stops...
Kind regards,
Jonas.
--
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Re: [asterisk-users] setting variable for a DID number
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tino Subject: [asterisk-users] setting variable for a DID number >Hello, >Is it possible to set a variable in dialpan when the someone calls a particular DID number so that i can use that variable for calls coming to that number only. As you asked the question: Exten => 5551212,1,Set(GLOBAL(TINO)=tino) Will set a variable to be used by any call when the user dials 5551212. If your incoming number is 5551212, you would want to use "ex-girlfriend" logic like this Exten => _X/5551212,n,Set(GLOBAL(TINO)=tino) Sets TINO when incoming line 5551212 rings. That's it for now. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] setting variable for a DID number
Hello, Is it possible to set a variable in dialpan when the someone calls a particular DID number so that i can use that variable for calls coming to that number only. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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
Ok. And how will we do for getting sip inbound calls from different ips and sending them to dahdi. 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
Re: [asterisk-users] CDR variables
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada Subject: Re: [asterisk-users] CDR variables >Ow... >I have =no commented, so I guess =yes is default?? >; Normally, CDR's are not closed out until after all extensions are finished >; executing. By enabling this option, the CDR will be ended before executing >; the "h" extension so that CDR values such as "end" and "billsec" may be >; retrieved inside of of this extension. >;endbeforehexten=no >So if I uncomment that, I will be able to use billsec in h exten... right? >Thanks Danny! As I read this you would have to uncomment and make yes as default is no. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR variables
Ow...
I have =no commented, so I guess =yes is default??
; Normally, CDR's are not closed out until after all extensions are finished
; executing. By enabling this option, the CDR will be ended before
executing
; the "h" extension so that CDR values such as "end" and "billsec" may be
; retrieved inside of of this extension.
;endbeforehexten=no
So if I uncomment that, I will be able to use billsec in h exten... right?
Thanks Danny!
On 18 August 2010 22:19, Danny Nicholas wrote:
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Tiago Geada
> *Subject:* [asterisk-users] CDR variables
>
>
>
> >Hello list!
>
> >I am trying to get hold of ${CDR(duration)} and ${CDR(billsec)} variables
> in h
>
> >It seems that these variables always return 0. I am using Asterisk
> version 1.6.2.11. Can't I get these values other than using CDR reccords ??
>
>
>
> In cdr.conf, is endbeforehexten=yes ?
>
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[asterisk-users] AMD message
Hello, Is there a way to capture the answering machine message when the dialer detects the answering machine. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] 3g call support for ISDN line
Dear All, i have a problem with 3g calling in asterisk with ISDN support . i tried it with the help of H324M gw . can any one tell that how i configure H324M gw . i fallow the bellow link http://www.voip-info.org/wiki/view/Asterisk+H324M http://sip.fontventa.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing with sipvicious ..
On Thu, 19 Aug 2010, Dana Harding wrote: > >> (I've just had 30GB of sipvicious traffic sent to my hosted servers in a >> 12-hour period - it came from what looked like a VPS host in France - >> trivially firewalled out, but even dropping the packets didn't stop the >> flood! It's so badly written it appears to just ignore any return codes >> that it doesn't want, or even no returns at all!) >> > http://blog.sipvicious.org/2010/06/how-to-crash-sipvicious-introducing.html > > It looks like it has been updated so that (with the newer version) this > won't happen. I doubt the new version will filter through for some months. People have a tool that appears to work for them, so they'll keep on using it. > I think that fail2ban or equivalent could be used to block the offending > IP, and also execute the provided svcrash.py which will send it's one > packet - possibly (if the attacker is using the older sipvicious) > stopping the traffic. To use svcrash, you need to identify the source port - and how many of the millions of people who're running asterisk via trixbox, innaflash, now, etc. actually know how to do that? (let alone get the SV sources and work out how to run them) > Of course that won't help if the attacker is not using sipvicious and > the other tool also ignores a lack of response. Indeed. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec choice
i do this by having 2 peers setup, one has a call limit of 10 and uses g729, the rest of the calls get sent to the second peer which uses ulaw. all calls attempt peer 1 if there's channels available it uses it if not it just moves through the dialplan to use the second one. On 19 August 2010 09:14, Deepika Nijhawan wrote: > Hi, > > > > Does anyone has an idea how to tell asterisk to use codec A for first 50 > calls and then codec B for rest of the calls. > > > > Thanks, > > Deepika > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every 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] Playing with sipvicious ..
> (I've just had 30GB of sipvicious traffic sent to my hosted servers in a > 12-hour period - it came from what looked like a VPS host in France - > trivially firewalled out, but even dropping the packets didn't stop the > flood! It's so badly written it appears to just ignore any return codes > that it doesn't want, or even no returns at all!) > http://blog.sipvicious.org/2010/06/how-to-crash-sipvicious-introducing.html It looks like it has been updated so that (with the newer version) this won't happen. I think that fail2ban or equivalent could be used to block the offending IP, and also execute the provided svcrash.py which will send it's one packet - possibly (if the attacker is using the older sipvicious) stopping the traffic. Of course that won't help if the attacker is not using sipvicious and the other tool also ignores a lack of response. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing with sipvicious ..
On 18/08/10 17:10, Gordon Henderson wrote: > > ... using it as a tool and understanding what it does... > > So one part of it's toolset identifys valid SIP accounts - and I was under > the impression that alwaysauthreject=yes was supposed to stop this... > > However, it sends a request for a highly probably non-existent account, > then sends requests for probably existing accounts and I guess compares > the results - account not found vs. bad username or password... It thus > trivially, and very quickly finds valid accounts when fed with a list of > accounts to try in the first place (e.g. 100-999, or 1000-, etc.) > > I wonder if it's time to introduce yet another parameter to it - which > will cause asterisk to return the same error code for all 3 conditions - > and return the "not found" error, even on bad username or password. > > It breaks the RFC even more, but might it be worth it? > > (I've just had 30GB of sipvicious traffic sent to my hosted servers in a > 12-hour period - it came from what looked like a VPS host in France - > trivially firewalled out, but even dropping the packets didn't stop the > flood! It's so badly written it appears to just ignore any return codes > that it doesn't want, or even no returns at all!) > > Gordon I've been playing with this a fair bit recently too, if only to gain myself a better understanding of the attacks so as to be able to prevent them better. I found that when sending Registers to Asterisk (I was testing with 1.4 since all my deployments are 1.4), alwaysauthreject does actually stop it from being able to determine real extension numbers. However I also found that making it send Invite requests means it can determine real extensions that are currently Registered. I've been using OSSEC to block source IPs that attacks come from. So far it seems to work well. Once you start silently dropping the inbound SIP traffic from the attacker they seem to go away very quickly (once the door is shut, no point them carrying on). I'm yet to see a more intelligent attack using distributed source IPs but I'm sure it'll happen. The scans I see happening usually come from random dynamic DSL addresses and the like from all over the place (inc within the UK) so I suspect these are virus infected zombies, so a distributed attack is surely easily possible. Something else I noticed is that once OSSEC is doing it's job (or whatever other automatic blocking script you use), the attacks stop. I have my systems set up to email me when an attack is blocked and after a few days, the attacks stop. Which I interpret as a sign that attackers are maintaining lists of known vulnerable IP addresses, which is common for things like ssh attacks, spam relays etc... I don't believe modifying Asterisk code to send non-RFC compliant relies is a good idea, I prefer the security layer to be handled by something else on top of Asterisk. I have also seen attacks exploiting bugs in Asterisk too, I'm not going to go into them here for obvious reasons but I guess these types of attacks will get more commonplace once people start getting a bit wiser to the current fairly basic port scan and extension enumeration attacks. cheers, Paul. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL Connect problem...
On Thu, Aug 19, 2010 at 4:27 AM, Geraint Lee wrote: > I would like to figure out why but can't really switch back now it "works" > since to replicate the problem... whatever it may be... i'd need to leave it > running live and wait for the live system to die... which obviously isn't > what i really want to happen :) > On 19 August 2010 08:11, Sherwood McGowan > wrote: >> >> On Wed, Aug 18, 2010 at 3:59 PM, Geraint Lee wrote: >> > This is what I ended up doing, working fine now. >> > Cheers >> > >> > On 18 August 2010 08:52, Nasir Iqbal wrote: >> >> >> >> Avoid to use MySQL dialplan application, instead write an AGI script >> >> for >> >> this purpose >> >> >> >> LOL, I hate to say this but writing an AGI script just adds yet >> another application layer to your total solution. OP, if you'd like to >> figure out WHY that was happening instead of abandoning the ship, I'd >> be glad to work with you to discover the cause. I've been using the >> MySQL Addon since the early days of ViaTalk back when 1.4 was still >> trunk code and the ARA was considered VERY experimental. I've never >> come across a problem with it that I couldn't figure out within a day >> so long as I stepped back and worked the "logical path" model of >> problem solving... >> >> Drop me a line, I think that I can figure it out within 20 questions >> and maybe a peek at a log ;-) >> >> 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 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > Totally understandable mate, trust me, I know how telecom/voip works..if it's working right now, DONT SCREW IT UP! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] MySQL Connect problem...
I would like to figure out why but can't really switch back now it "works" since to replicate the problem... whatever it may be... i'd need to leave it running live and wait for the live system to die... which obviously isn't what i really want to happen :) On 19 August 2010 08:11, Sherwood McGowan wrote: > On Wed, Aug 18, 2010 at 3:59 PM, Geraint Lee wrote: > > This is what I ended up doing, working fine now. > > Cheers > > > > On 18 August 2010 08:52, Nasir Iqbal wrote: > >> > >> Avoid to use MySQL dialplan application, instead write an AGI script for > >> this purpose > >> > > LOL, I hate to say this but writing an AGI script just adds yet > another application layer to your total solution. OP, if you'd like to > figure out WHY that was happening instead of abandoning the ship, I'd > be glad to work with you to discover the cause. I've been using the > MySQL Addon since the early days of ViaTalk back when 1.4 was still > trunk code and the ARA was considered VERY experimental. I've never > come across a problem with it that I couldn't figure out within a day > so long as I stepped back and worked the "logical path" model of > problem solving... > > Drop me a line, I think that I can figure it out within 20 questions > and maybe a peek at a log ;-) > > 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 > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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
On Thu, Aug 19, 2010 at 4:11 AM, Nasir Iqbal wrote:
> little syntax mistake, try this
> exten => s,1,Dial(SIP/Ext400&Local/${ext...@home-context)
> [home-context]
> exten => s,1,Set(CALLERID(num)=44112233445566)
> exten => s,n,Dial(SIP/TheWorld/441234567890)
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Noo Please let's not go down that nasty trail of using
local channels...ugh...besides, there's still no conceivable reason
why one would need to do anything out of the ordinary, the party being
called outside the office will have to be sent through the provider,
in which case the external CLID should be defined already
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Re: [asterisk-users] Calling Line Identity - any ideas
little syntax mistake, try this
exten => s,1,Dial(SIP/Ext400&Local/${ext...@home-context)
[home-context]
exten => s,1,Set(CALLERID(num)=44112233445566)
exten => s,n,Dial(SIP/TheWorld/441234567890)
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Re: [asterisk-users] Calling Line Identity - any ideas
Hi,
here is a trick for you!
exten => s,1,Dial(SIP/Ext400&Local/${EXTEN}/home-context)
[home-context]
exten => s,1,Set(CALLERID(num)=44112233445566)
exten => s,1,Dial(SIP/TheWorld/441234567890)
Regards
On Thu, Aug 19, 2010 at 12:21 PM, Paddy Grice wrote:
> Hi list
>
> I have a requirement that I just don't know how to address - I don't think
> its strange but can't find any pointers anywhere.
>
> I have a user that wishes to have a "multi phone" divert. By that I mean
>
> "calls made to his extension say Ext200 can be redirected to a different
> extension say Ext400 and also to his home landline.
>
> Doing the dial is fine using Dial(SIP/Ext400&SIP/TheWorld/441234567890)
>
> The problem is CLID -
>
> At the moment internal calls (Ext to Ext) show a CLID "EXTxxx" and External
> Calls show the received CLID.
>
> When the phone is redirected to both Internal and external numbers he wants
> the correct CLI displayed on both phones.
>
> So with the redirect operational
>
> 1) a call from the outside world to his DID number will show the received
> CLI(ANI) on both devices - this works
> BUT
> 2) a call from an office extension needs to show "EXTxxx" on the extension
> (Ext400) but show the office telephone number on the landline
>
> so in pseudo code I want to do something like
>
> Dial ( SIP/Ext400 using CLID "EXT123" & SIP/TheWorld/441234567890 using
> CLID "44112233445566" )
>
> Any ideas ?
>
> Paddy
>
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--
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ICT Innovations
http://www.ictinnovations.com/
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Re: [asterisk-users] Codec choice
On Thu, Aug 19, 2010 at 3:37 AM, Steve Howes wrote: > > On 19 Aug 2010, at 09:14, Deepika Nijhawan wrote: >> Does anyone has an idea how to tell asterisk to use codec A for first 50 >> calls and then codec B for rest of the calls. > > You could create two separate trunks, one for each codec? > > S > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > Good Point! LOL, I went with a much more complicated method...sleep deprivation at it's finest perhaps? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 19 Aug 2010, at 09:14, Deepika Nijhawan wrote: > Does anyone has an idea how to tell asterisk to use codec A for first 50 > calls and then codec B for rest of the calls. You could create two separate trunks, one for each codec? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec choice
On Thu, Aug 19, 2010 at 3:14 AM, Deepika Nijhawan wrote: > Hi, > > > > Does anyone has an idea how to tell asterisk to use codec A for first 50 > calls and then codec B for rest of the calls. > > > > Thanks, > > Deepika > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every 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 I can think of is to use a global variable that you increment each time a new call spawns, and once it's over your threshold (50 in this case) use the CHANNEL() function to set the audio format to the codec you want (google voip-info function CHANNEL) 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] Calling Line Identity - any ideas
I'll see if I can make it a little clearer for you... When ext 123 makes a call to another SIP device through the server but not requiring a middleman (i.e. a third party provider that allows you to dial to parties outside the immediate control of your PBX) to accomplish the call, what happens is Asterisk sets the CallerID on leg B of the call (the leg that's "going" to the other internal extension) to Ext 123's configured CLID. However, when you call through a provider (the "middleman"), the call has to use a peer/friend definition OTHER than the original one (Ext 123's)...it has to perform the call via the friend account you made for ABCTelco, which could have included the CallerID information you wanted displayed over all calls being sent outbound through that provider... Here's a really simple way to view this... In the dialplan, the call to another internal extension is dialed thus: Dial(SIP/456) However, when you're dialing through a peer (middleman, provider, whatever) it looks like this: Dial(SIP/3148083...@provider) See, the @ indicates that you want asterisk to use another account as a trunk to reach the number you're dialing. In your dialplan, when you dial the outside world, what's after the @? Whatever it is, go to that entry in your sip.conf, and you'll be 90% of the way home On Thu, Aug 19, 2010 at 2:54 AM, Paddy Grice wrote: > Hi Sherwood > > Maybe my miss-understanding sip.conf I will try and see what happens - but I > don't understand how "sip" would know which CLID to send to each sip > endpoint - internal or external. > > BTW this is all to get return of missed calls working. > > 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. > > Will check up on sip.conf > > Paddy > > -Original Message- > From: Sherwood McGowan [mailto:sherwood.mcgo...@gmail.com] > Sent: 19 August 2010 08:35 > To: pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial > Discussion > Subject: Re: [asterisk-users] Calling Line Identity - any ideas > > I'd have to say off the top of my head that this should already work as long > as the trunk you're sending calls to the outside world over has the CallerID > setting set AND probably sendrpid=yes...in the sip configuration for both of > those items...past that, I could dig a bit > > Cheers, > Sherwood McGowan > > On Thu, Aug 19, 2010 at 2:25 AM, Paddy Grice wrote: >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paddy >> Grice >> Sent: 19 August 2010 08:21 >> To: asterisk-users@lists.digium.com >> Subject: [asterisk-users] Calling Line Identity - any ideas >> >> Hi list >> >> I have a requirement that I just don't know how to address - I don't >> think its strange but can't find any pointers anywhere. >> >> I have a user that wishes to have a "multi phone" divert. By that I >> mean >> >> "calls made to his extension say Ext200 can be redirected to a >> different extension say Ext400 and also to his home landline. >> >> Doing the dial is fine using >> Dial(SIP/Ext400&SIP/TheWorld/441234567890) >> >> The problem is CLID - >> >> At the moment internal calls (Ext to Ext) show a CLID "EXTxxx" and >> External Calls show the received CLID. >> >> When the phone is redirected to both Internal and external numbers he >> wants the correct CLI displayed on both phones. >> >> So with the redirect operational >> >> 1) a call from the outside world to his DID number will show the >> received >> CLI(ANI) on both devices - this works >> BUT >> 2) a call from an office extension needs to show "EXTxxx" on the >> extension >> (Ext400) but show the office telephone number on the landline >> >> so in pseudo code I want to do something like >> >> Dial ( SIP/Ext400 using CLID "EXT123" & SIP/TheWorld/441234567890 >> using CLID "44112233445566" ) >> >> Any ideas ? >> >> Paddy >> >> >> >> Forgot to say - I am using Version 1.4.33.1 >> >> Paddy >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every 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, Does anyone has an idea how to tell asterisk to use codec A for first 50 calls and then codec B for rest of the calls. Thanks, Deepika -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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
Hi Sherwood Maybe my miss-understanding sip.conf I will try and see what happens - but I don't understand how "sip" would know which CLID to send to each sip endpoint - internal or external. BTW this is all to get return of missed calls working. 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. Will check up on sip.conf Paddy -Original Message- From: Sherwood McGowan [mailto:sherwood.mcgo...@gmail.com] Sent: 19 August 2010 08:35 To: pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calling Line Identity - any ideas I'd have to say off the top of my head that this should already work as long as the trunk you're sending calls to the outside world over has the CallerID setting set AND probably sendrpid=yes...in the sip configuration for both of those items...past that, I could dig a bit Cheers, Sherwood McGowan On Thu, Aug 19, 2010 at 2:25 AM, Paddy Grice wrote: > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paddy > Grice > Sent: 19 August 2010 08:21 > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Calling Line Identity - any ideas > > Hi list > > I have a requirement that I just don't know how to address - I don't > think its strange but can't find any pointers anywhere. > > I have a user that wishes to have a "multi phone" divert. By that I > mean > > "calls made to his extension say Ext200 can be redirected to a > different extension say Ext400 and also to his home landline. > > Doing the dial is fine using > Dial(SIP/Ext400&SIP/TheWorld/441234567890) > > The problem is CLID - > > At the moment internal calls (Ext to Ext) show a CLID "EXTxxx" and > External Calls show the received CLID. > > When the phone is redirected to both Internal and external numbers he > wants the correct CLI displayed on both phones. > > So with the redirect operational > > 1) a call from the outside world to his DID number will show the > received > CLI(ANI) on both devices - this works > BUT > 2) a call from an office extension needs to show "EXTxxx" on the > extension > (Ext400) but show the office telephone number on the landline > > so in pseudo code I want to do something like > > Dial ( SIP/Ext400 using CLID "EXT123" & SIP/TheWorld/441234567890 > using CLID "44112233445566" ) > > Any ideas ? > > Paddy > > > > Forgot to say - I am using Version 1.4.33.1 > > Paddy > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every 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] Directory routing to wrong extension if dial tones are pressed too quick.
Are your calls coming in over IAX, SIP, or DAHDI/ZAP? this will make a big difference in the way I'd go about trying to figure out how to resolve the issue. If it's SIP, try doing a network capture of all SIP and RTP traffic in and out of the server and then try to replicate the problem. also, turn on DTMF logging, and check things like sip.conf's relaxdtmf setting and what also what dtmfmode you're using as this can both make a HUGE difference... On Tue, Aug 17, 2010 at 2:43 PM, Eddie Mikell wrote: > Hi All, > > Have completely moved off the old ESI system, and things have been going > pretty good with the new server. > > I have one issue, which has been reported by several of our customers. > I've tested it, and it does indeed seem to be a problem. > > When the customer is asked to dial in the first three letters of the > person they are trying to reach, they will be routed to the wrong > extension. > > The problem seems to revolve around how quickly the keys are pressed. > So if 645 for Mikell is pressed very quickly, they end up being routed > to Sarah Fish. But if they take their time, say 2 seconds between each > keystroke, everything works ok. > > Are they any settings that can be adjusted for this? > > Thanks, > Eddie Mikell > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every 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
I'd have to say off the top of my head that this should already work as long as the trunk you're sending calls to the outside world over has the CallerID setting set AND probably sendrpid=yes...in the sip configuration for both of those items...past that, I could dig a bit Cheers, Sherwood McGowan On Thu, Aug 19, 2010 at 2:25 AM, Paddy Grice wrote: > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paddy Grice > Sent: 19 August 2010 08:21 > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Calling Line Identity - any ideas > > Hi list > > I have a requirement that I just don't know how to address - I don't think > its strange but can't find any pointers anywhere. > > I have a user that wishes to have a "multi phone" divert. By that I mean > > "calls made to his extension say Ext200 can be redirected to a different > extension say Ext400 and also to his home landline. > > Doing the dial is fine using Dial(SIP/Ext400&SIP/TheWorld/441234567890) > > The problem is CLID - > > At the moment internal calls (Ext to Ext) show a CLID "EXTxxx" and External > Calls show the received CLID. > > When the phone is redirected to both Internal and external numbers he wants > the correct CLI displayed on both phones. > > So with the redirect operational > > 1) a call from the outside world to his DID number will show the received > CLI(ANI) on both devices - this works > BUT > 2) a call from an office extension needs to show "EXTxxx" on the extension > (Ext400) but show the office telephone number on the landline > > so in pseudo code I want to do something like > > Dial ( SIP/Ext400 using CLID "EXT123" & SIP/TheWorld/441234567890 using CLID > "44112233445566" ) > > Any ideas ? > > Paddy > > > > Forgot to say - I am using Version 1.4.33.1 > > Paddy > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every 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] realtime sip peers : musiconhold class
Converting with sox works well as followed : sox -V intro.wav -c 1 -r 8000 intro2.wav To convert with asterisk convert, I needed to use an absolute path : asterisk -rx "file convert /var/lib/asterisk/moh/folder/intro.wav /var/lib/asterisk/folder/intro.alaw" All works well. Jonas. On 08/19/2010 02:31 AM, Nasir Iqbal wrote: Hi to convert wav file use following sox 'orgFile' -w -r 8000 -c 1 -s 'fixedFile' while replace orgFile and fixedFile with actual filenames If still now luck try with mp3 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
Re: [asterisk-users] Calling Line Identity - any ideas
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paddy Grice Sent: 19 August 2010 08:21 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Calling Line Identity - any ideas Hi list I have a requirement that I just don't know how to address - I don't think its strange but can't find any pointers anywhere. I have a user that wishes to have a "multi phone" divert. By that I mean "calls made to his extension say Ext200 can be redirected to a different extension say Ext400 and also to his home landline. Doing the dial is fine using Dial(SIP/Ext400&SIP/TheWorld/441234567890) The problem is CLID - At the moment internal calls (Ext to Ext) show a CLID "EXTxxx" and External Calls show the received CLID. When the phone is redirected to both Internal and external numbers he wants the correct CLI displayed on both phones. So with the redirect operational 1) a call from the outside world to his DID number will show the received CLI(ANI) on both devices - this works BUT 2) a call from an office extension needs to show "EXTxxx" on the extension (Ext400) but show the office telephone number on the landline so in pseudo code I want to do something like Dial ( SIP/Ext400 using CLID "EXT123" & SIP/TheWorld/441234567890 using CLID "44112233445566" ) Any ideas ? Paddy Forgot to say - I am using Version 1.4.33.1 Paddy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Calling Line Identity - any ideas
Hi list I have a requirement that I just don't know how to address - I don't think its strange but can't find any pointers anywhere. I have a user that wishes to have a "multi phone" divert. By that I mean "calls made to his extension say Ext200 can be redirected to a different extension say Ext400 and also to his home landline. Doing the dial is fine using Dial(SIP/Ext400&SIP/TheWorld/441234567890) The problem is CLID - At the moment internal calls (Ext to Ext) show a CLID "EXTxxx" and External Calls show the received CLID. When the phone is redirected to both Internal and external numbers he wants the correct CLI displayed on both phones. So with the redirect operational 1) a call from the outside world to his DID number will show the received CLI(ANI) on both devices - this works BUT 2) a call from an office extension needs to show "EXTxxx" on the extension (Ext400) but show the office telephone number on the landline so in pseudo code I want to do something like Dial ( SIP/Ext400 using CLID "EXT123" & SIP/TheWorld/441234567890 using CLID "44112233445566" ) Any ideas ? Paddy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] MySQL Connect problem...
On Wed, Aug 18, 2010 at 3:59 PM, Geraint Lee wrote: > This is what I ended up doing, working fine now. > Cheers > > On 18 August 2010 08:52, Nasir Iqbal wrote: >> >> Avoid to use MySQL dialplan application, instead write an AGI script for >> this purpose >> LOL, I hate to say this but writing an AGI script just adds yet another application layer to your total solution. OP, if you'd like to figure out WHY that was happening instead of abandoning the ship, I'd be glad to work with you to discover the cause. I've been using the MySQL Addon since the early days of ViaTalk back when 1.4 was still trunk code and the ARA was considered VERY experimental. I've never come across a problem with it that I couldn't figure out within a day so long as I stepped back and worked the "logical path" model of problem solving... Drop me a line, I think that I can figure it out within 20 questions and maybe a peek at a log ;-) 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
