Re: [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?
On Thu, May 20, 2010 at 7:09 PM, Leif Madsen wrote: > Olivier wrote: > > As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an > > issue with BLF-pickup which kept me from going further. > > Which bug number have you reported your issue in? > > Leif. > > I am using it because I needed reliable T.38 and opted tof Fax for Asterisk. Asterisk segfaults regularly. If 1.2 or 1.4 do what you want, then forget 1.6 unitl it mature. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] file command with alaw file
This may be totally irrelevant and it may send you down the wrong track, but I thought I would mention it: There is a bug which can prevent recent versions of asterisk from creating proper headers in WAV files. The bug shows up on Solaris systems but Linux is theoretically not immune to it. If you are creating raw ulaw or alaw files you should probably not expect any headers when things are working normally. But if you are creating wav files (which can contain alaw or ulaw data) and you find a bunch of null characters (0x0) in the header, then you may have hit the bug. See: https://issues.asterisk.org/view.php?id=16610 > I did check content of two alaw files (using a hex editor) generated > from two aboves cases. For the one fromo CentOS 5.2, beginning bytes > look likes : > > RIFF1^0.WAVEfmt at @...fact.^0.data.^0... > > and the one from CentOS 5.5 > > ..RQVTVXEMBAX > > It seem like the first one have some information about file format, > which make our convert tool works correctly, and which the second one > doesnt have. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Softphones on thin clients...
Not open source, nor free...but certainly available. --Original Message Text--- From: bruce bruce Date: Thu, 20 May 2010 15:33:41 -0400 Is the Java soft phone an open source or obtainable? I am just checking their site and it seems they only provide service??!! Their java web based client is built neatly. Would like to test that on my servers. On Thu, May 20, 2010 at 3:21 PM, wrote: I've used HP Thin Clients as embedded hosts for Asterisk. The T5700 models that I have are 1 GHz CPUs, more recent models should be able to run a soft phone without too much trouble. They all have local USB ports, making USB headsets as good solution. Another alternative might be to used a soft phone implemented as a web plug-in or activex object. Tim Panton of PhoneFromHere.com has a great Java soft phone object that we use to make G.722 calls to the ZipDX conference bridge for the VoIP Users Conference every week. Michael Graves mgraves mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgra...@mstvp.onsip.com skype mjgraves > Original Message > Subject: Re: [asterisk-users] Softphones on thin clients... > From: Carlos Chavez > Date: Thu, May 20, 2010 1:36 pm > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > > -Original Message- > > From: asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes > > Sent: Thursday, May 20, 2010 1:51 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Softphones on thin clients... > > > > > > On 20 May 2010, at 18:35, Carlos Chavez wrote: > > > I am worried about conflicts when running 10 softphones on the same > > > server since they will all try to use por 5060. > > > > And the fact most terminal services servers/clients still don't support > > audio input.. only output.. > > Since the little box has a MIC jack I suppose that it should support > audio input. These boxes will be running Windows and using Eyebeam. > > -- > Telecomunicaciones Abiertas de México S.A. de C.V. > Carlos Chávez Prats > Director de Tecnología > +52-55-91169161 ext 2001-- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every 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 -- Michael Graves mgravesmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] OT - Which SIP hardphone with 100 BLF ?
On Thu, 20 May 2010, Gordon Henderson wrote: > On Thu, 20 May 2010, SIP wrote: > >> Even IF you could get a keyboard with lights you could individually turn >> on and off (never seen one), > > http://www.artlebedev.com/everything/optimus/ > > Bit expensive though... > > Gordon > Heh. A $2400 keyboard. That's crazy. Cool though. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] DAHDI and ESXi
On Thu, May 20, 2010 at 7:14 PM, Alec Davis wrote: > The following link may be a suitable workaround > > How do I change the type of line from E1 to T1/J1 without using jumpers? > http://kb.digium.com/entry/121/ > Alec, Thank you, thats worked for me. Although, the 'insmod wct4xxp t1e1override=0xFF' did not work exactly, but I managed to add 'options wct4xxp t1e1override=0' to /etc/modprobe.d/dahdi to achieve the same result. Now onto testing this Conor -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Cause and cure for "Exceptionally long voice queue length queuing to Local"?
David Cunningham wrote: > Hello, > > We're seeing lots of warnings like the following, running Asterisk > 1.6.1.12. Does anyone know the cause or cure? > > One explanation I've come across is that the peer is congested and > sending RTCP messages asking us to slow the RTP down. Is there any way > we can verify this? > > [May 17 13:42:45] WARNING[27482] channel.c: Exceptionally long voice > queue length queuing to Local/12126412...@asterisk-phone-7e3d;1 > > Thanks in advance! Try upgrading to 1.6.1.13. You're using a version of Asterisk from early December 2009. Doing a search for closed issues on the Asterisk issue tracker at https://issues.asterisk.org caused me to find bug 15609 (https://issues.asterisk.org/view.php?id=15609) which was committed on December 30, 2009. On January 11, 2010, Asterisk version 1.6.1.13-rc1 was created which contains the commit from December 30, 2009, and was subsequently released as 1.6.1.13 on January 15, 2010. The ChangeLog showing the commit is here: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.13 The release is available here: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Which issue is keeping you from updrading to 1.6.2 ?
Greg Woods wrote: > I am still running 1.4 because of this bug: > > https://issues.asterisk.org/view.php?id=15129 > > I haven't tried any 1.6 versions recently; looks like some patches have > been checked in since I last tried it, although the bug is not closed. > So I may have to try it again when I get some time. Is that a typo? That bug has been closed since September 20, 2009. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Which issue is keeping you from updrading to1.6.2 ?
Danny Nicholas wrote: > If I'm going to bother with 1.6.2, I'll wait a few months for 1.8. But in > the spirit of your question: > (1) dialplan conversion > (2) loss of functions like Gosub Can you be more specific about what 1) and 2) mean? Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Which issue is keeping you from updrading to 1.6.2 ?
Olivier wrote: > As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an > issue with BLF-pickup which kept me from going further. Which bug number have you reported your issue in? Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Which issue is keeping you from updrading to 1.6.2 ?
David Backeberg wrote: > meetme CLI arguments changed between 1.6.0 and 1.6.2 > Don't know where the delta was, and I haven't looked. > I prefer the new syntax, and especially prefer the 'concise' option, > but it might break features people have built in the past. > > Specifically, > 1.6.0 'meetme' is replaced with 1.6.2 'meetme list' > 1.6.2 'meetme' errors out, and requires an argument > 1.6.0 'meetme list' would error out wanting a room argument > 'meetme list ' works same on 1.6.0 and 1.6.2 > 'meetme list concise' and 'meetme list concise' in 1.6.2 are nice. This sounds like a solution for cli_aliases.conf. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] file command with alaw file
On Thu, May 20, 2010 at 1:49 AM, Pham Quy wrote: > Hi, > > How can I convert FROM ALAW file, which generated by asterisk apps > (monitor, or record app), to format (wav or mp3) that is playable by > music player?? Can Sox do this? > >From alaw to wav, you can use Asterisk's CLI f" file convert /var/lib/asterisk/sounds/soundfile.alaw /var/lib/asterisk/sounds/soundfile.wav to go from alaw to mp3, first convert to wav, then use lame /var/lib/asterisk/sounds/soundfile.wav /var/lib/asterisk/sounds/soundfile.mp3 sox looks like it can ogg/vorbis, but mine doesn't list mp3. You might fetch the source for sox and see if it can do mp3; lame is probably just as easy to obtain and use. murf > > I have an asterisk 1.6.2.6 on my CentOS 5.2. I record audio clip by > mixmonitor app and use file command to check the alaw output, and here > is output: > > - > 983006584-20100517-125002.alaw: RIFF (little-endian) data, WAVE audio, > ITU G.711 A-law, mono 8000 Hz > - > > How could file command recognize the format as there is no header in the > output file? Or Did I probably miss something making asterisk yield > incorrect alaw files? > > Please help, thanks > > Quyps > > On Thu, 2010-05-20 at 00:50 -0600, Steve Murphy wrote: > > Quyps-- > > > > I've noticed in general that the ulaw, alaw, gsm, slin files used and > > generated by > > asterisk do not have headers (the RIFF stuff), and asterisk is not > > expecting them. in general they > > will louse up Asterisk's ability to play the sound. They are just raw > > data and the extension > > on the file name (.gsm, or .ulaw, etc) is the only clue to the file > > format/contents. > > > > In general, if you need a sound file of your own making in a certain > > format, you can convert > > to most of the formats using sox in linux, but really, the best thing > > to do is record the source > > sound file in cd-quality sound WAV format, in 44 khz sampling rate, or > > higher, and then > > use sox to convert to 8khz format. Asterisk can do some of this via > > the file convert CLI > > command, ( on the asterisk cli, type "help file convert"). You'd have > > to judge for yourself > > if "file convert tt-weasels.gsm tt-weasels.ulaw" which would convert > > the 8khz gsm format to > > 8 khz ulaw, or "sox -v 0.7 tt-weasels.44khz.wav -r 8000 -c 1 -t sw > > tt-weasels.raw;" > > "sox -r 8000 -c 1 -t sw tt-weasels.raw -t ul tt-weasels.ulaw" which > > is the way the Asterisk > > sounds are produced from the the cd-quality sounds. They would seem a > > bit equivalent. > > > > I wonder if just "sox -v 0.7 tt-weasels.44khz.wav -r 8000 -c 1 -t ul > > tt-weasels.ulaw" would > > sound any better... you audio engineers out there may have an opinion. > > > > I've personally noted that not all linux distributions provide the > > same version of sox; > > some distribute sox with an absolute minimum of sound formats > > built-in. You may have > > to go out and find all the libraries and roll your own sox. > > > > murf > > > > > > > > > > > > On Wed, May 19, 2010 at 10:34 PM, Pham Quy wrote: > > On Mon, 2010-05-17 at 17:49 +0700, Pham Quy wrote: > > > On Mon, 2010-05-17 at 13:06 +0700, Pham Quy wrote: > > > > hi all, > > > > > > > > I install Asterisk 1.6 on Centos 5.2 (kernel 2.6.18-92.el5 > > #1 SMP Tue > > > > Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux) > > and do record > > > > audio clip with mixmonitor() as alaw file (softphone is > > also configured > > > > with alaw active only). Using file command i can get the > > following > > > > information > > > > > > > > 983006584-20100517-125002.alaw: RIFF (little-endian) data, > > WAVE audio, > > > > ITU G.711 A-law, mono 8000 Hz > > > > > > > > But when i install the same system on Centos 5.5 (kernel > > 2.6.18-92.el5 > > > > #1 SMP Tue Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 > > GNU/Linux) i > > > > could get the same information with file command. File > > command > > > > recognized alaw file as JPEG image: > > > > > > > > 983006584-20100517-123825.alaw: JPEG image data > > > > > > > > I guess i may miss something when i setup the new on on > > Centos 5.5, but > > > > u dont know what it is. Anyone have idea about this? > > > > > > > > please help. > > > > > > > > Thanks in advance. > > > > Quyps > > > > > > I did check content of two alaw files (using a hex editor) > > generated > > > from two aboves cases. For the one fromo CentOS 5.2, > > beginning bytes > > > look likes : > > > > > > riff1^0.wavefmt@...@...fact.^0.data.^0... > > > > > > and the one from CentOS 5.5 > > > > > > ..RQVTVX
Re: [asterisk-users] Which issue is keeping you from updrading to 1.6.2 ?
On Thu, 2010-05-20 at 17:41 +0200, Olivier wrote: > Hi, > > I'm evaluating what could keep me from upgrading production systems to > 1.6.2. I am still running 1.4 because of this bug: https://issues.asterisk.org/view.php?id=15129 I haven't tried any 1.6 versions recently; looks like some patches have been checked in since I last tried it, although the bug is not closed. So I may have to try it again when I get some time. --Greg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.0.28 and 1.6.1.20 Now Available
The Asterisk Development Team has announced the final maintenance releases of Asterisk branches 1.6.0 and 1.6.1 as versions 1.6.0.28 and 1.6.1.20. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The Asterisk releases for 1.6.0.28 and 1.6.1.20 are the last maintenance releases for Asterisk branches 1.6.0 and 1.6.1 and have now moved to security maintenance only. The releases of Asterisk 1.6.0.28 and 1.6.1.20 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community developers: * Fix issue where MixMonitor() recordings would be shorter than total duration. (Closes issue #17078. Reported,tested by geoff2010. Patched by dhubbard) * When StopMonitor() is called, ensure it will not be restarted by a channel event. (Closes issue #16590. Reported, patched by kkm) * Allow hidecalleridname feature to work. (Closes issue #17143. Reported, patched by djensen99) * Resolve deadlocks in chan_local. (Closes issue #17185. Reported, tested by schmoozecom, GameGamer43) * Ensure channel state is not incorrectly set in the case of a very early answer by chan_dahdi. (Closes issue #17067. Reported, patched by tzafrir) * Registration fix for SIP realtime. Make sure realtime fields are not empty. (Closes issue #17266. Reported, patched by Nick_Lewis. Tested by sberney) More information about the changes to maintenance support can be found at: http://www.asterisk.org/node/49924 Information about the Asterisk maintenance schedule is available at: http://www.asterisk.org/asterisk-versions For a full list of changes in the current release candidates, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.28 http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.20 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ?
On Thu, 20 May 2010, SIP wrote: >Even IF you could get a keyboard with lights you could individually turn >on and off (never seen one), http://www.artlebedev.com/everything/optimus/ Bit expensive though... 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] Softphones on thin clients...
Is the Java soft phone an open source or obtainable? I am just checking their site and it seems they only provide service??!! Their java web based client is built neatly. Would like to test that on my servers. On Thu, May 20, 2010 at 3:21 PM, wrote: > I've used HP Thin Clients as embedded hosts for Asterisk. The T5700 > models that I have are 1 GHz CPUs, more recent models should be able to > run a soft phone without too much trouble. They all have local USB > ports, making USB headsets as good solution. > > Another alternative might be to used a soft phone implemented as a web > plug-in or activex object. Tim Panton of PhoneFromHere.com has a great > Java soft phone object that we use to make G.722 calls to the ZipDX > conference bridge for the VoIP Users Conference every week. > > Michael Graves > mgraves mstvp.com > o(713) 861-4005 > c(713) 201-1262 > sip:mjgra...@mstvp.onsip.com > skype mjgraves > > > Original Message > > Subject: Re: [asterisk-users] Softphones on thin clients... > > From: Carlos Chavez > > Date: Thu, May 20, 2010 1:36 pm > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > > > > > > -Original Message- > > > From: asterisk-users-boun...@lists.digium.com > > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve > Howes > > > Sent: Thursday, May 20, 2010 1:51 PM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: Re: [asterisk-users] Softphones on thin clients... > > > > > > > > > On 20 May 2010, at 18:35, Carlos Chavez wrote: > > > > I am worried about conflicts when running 10 softphones on the same > > > > server since they will all try to use por 5060. > > > > > > And the fact most terminal services servers/clients still don't support > > > audio input.. only output.. > > > > Since the little box has a MIC jack I suppose that it should > support > > audio input. These boxes will be running Windows and using Eyebeam. > > > > -- > > Telecomunicaciones Abiertas de México S.A. de C.V. > > Carlos Chávez Prats > > Director de Tecnología > > +52-55-91169161 ext 2001-- > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > >http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Asterisk-Users] Asterisk transfer to a conference using feature code?
Is it possible to use an Asterisk feature code to transfer a call to a specific extension? For instance, if you take a call, and the caller wants to go to a conference, it would be nice to use a feature code for this, rather than going through a longer transfer sequence. e.g.: - You have a meetme conference: [conferences] exten => 21,1,NoOp(MeetMe Conference) exten => 21,n,MeetMe(50,pM) ;p=prompt for pin, M=music for first caller exten => 21,n,Hangup - You then want to define a feature code *5 in features.conf which will blind transfer the caller to (conferences,21,1) Any suggestions/examples as to how to set this up? 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] Softphones on thin clients...
I've used HP Thin Clients as embedded hosts for Asterisk. The T5700 models that I have are 1 GHz CPUs, more recent models should be able to run a soft phone without too much trouble. They all have local USB ports, making USB headsets as good solution. Another alternative might be to used a soft phone implemented as a web plug-in or activex object. Tim Panton of PhoneFromHere.com has a great Java soft phone object that we use to make G.722 calls to the ZipDX conference bridge for the VoIP Users Conference every week. Michael Graves mgraves mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgra...@mstvp.onsip.com skype mjgraves > Original Message > Subject: Re: [asterisk-users] Softphones on thin clients... > From: Carlos Chavez > Date: Thu, May 20, 2010 1:36 pm > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > > -Original Message- > > From: asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes > > Sent: Thursday, May 20, 2010 1:51 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Softphones on thin clients... > > > > > > On 20 May 2010, at 18:35, Carlos Chavez wrote: > > > I am worried about conflicts when running 10 softphones on the same > > > server since they will all try to use por 5060. > > > > And the fact most terminal services servers/clients still don't support > > audio input.. only output.. > > Since the little box has a MIC jack I suppose that it should support > audio input. These boxes will be running Windows and using Eyebeam. > > -- > Telecomunicaciones Abiertas de México S.A. de C.V. > Carlos Chávez Prats > Director de Tecnología > +52-55-91169161 ext 2001-- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every 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] Checking blank CallerID in Dialplan
You will find there are an infinite number of bogus CLID's that these scumbags use to thwart screening. Such things as invalid NPA, invalid office code are common. Blank is seldom used any more. Here in the US at least, even with the do not call list ( federal ) and various state do not call lists, they still squeek through. Part of the stupidity of the DNC list ( federal ) is that there is a charge to USE the list so the bottom feeders don't bother, and with some providers allowing phony CLID there is little hope. What we do is screen for the private, blocked, etc, then capture the number and name into a MySQL database. A php script then can display that, and allow for setting a field to block if it ever comes through again. In the dialplan, if the number comes through but NO name, they go to a VM box that allow legit callers to leave a message, the rest simply hang up. Even valid numbers of callers we want no conversation with can be blocked as well. The script also allows entry of known undesirables to be entered. Once a number is blocked, they are routed to an intercept recording that APPEARS to be authentic, and often that is enough to discourage further calling. Also be prepared for the infrequent call that is missing any CLID. You are only limited by your imagination, and the undesirable callers tenacity to outfox your scripts. Be prepared to continue to change and improve your efforts. John Novack Myles Wakeham wrote: > I am trying to implement a change to our Dialplan that will thwart > tele-spammers that are calling us with blanked out caller ID. > > The caller IDs seem to vary between originating callers when they block > caller ID. I've seen the following: > > "anonymous" > "" > > So I'm checking for these. However recently one company seems to be > bypassing this, so what I wanted to do was implement some logic that > checks for actual numbers in the caller ID. > > We have a couple of different SIP providers for incoming calls. Some > prefix numbers with a + and others don't. But I'm logging incoming > calls that are getting through our tele-spam filter and it seems that > they are blank, but I suspect they contain empty spaces which is why our > matches don't work. > > Does anyone have some sample DialPlan code that they are using to thwart > incoming calls with no caller ID? I was thinking of maybe converting > the caller ID num to a numeric value and testing for 'not equal to 0' > but that won't work with the + prefix. > > All suggestions greatly appreciated. > > Myles > > > > > > Checked by AVG - www.avg.com > Version: 9.0.819 / Virus Database: 271.1.1/2884 - Release Date: 05/19/10 > 14:26:00 > > -- Dog is my co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which issue is keeping you from updrading to1.6.2 ?
If I'm going to bother with 1.6.2, I'll wait a few months for 1.8. But in the spirit of your question: (1) dialplan conversion (2) loss of functions like Gosub -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Thursday, May 20, 2010 11:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Which issue is keeping you from updrading to1.6.2 ? On Thu, May 20, 2010 at 11:41 AM, Olivier wrote: > Hi, > > I'm evaluating what could keep me from upgrading production systems to > 1.6.2. > As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an > issue with BLF-pickup which kept me from going further. > > Have you met other issues I should include include in my checklist ? meetme CLI arguments changed between 1.6.0 and 1.6.2 Don't know where the delta was, and I haven't looked. I prefer the new syntax, and especially prefer the 'concise' option, but it might break features people have built in the past. Specifically, 1.6.0 'meetme' is replaced with 1.6.2 'meetme list' 1.6.2 'meetme' errors out, and requires an argument 1.6.0 'meetme list' would error out wanting a room argument 'meetme list ' works same on 1.6.0 and 1.6.2 'meetme list concise' and 'meetme list concise' in 1.6.2 are nice. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Checking blank CallerID in Dialplan
This is a drawn-out, but efficient way to "fix" this problem. Create two
programs. Program 1 reads Master.csv (or whatever you use to store your CDR
in). Reads through CDR and creates a "blacklist" of numbers and ID's. write
blacklist to a text file or database. Program 2 runs from dialplan as AGI.
Returns "Blacklist" or "OK" to dialplan as local variable. I'm a "Perl
Weenie", so I would do this with text files, but C code and a database would
be more efficient (actually, unless the list got quite cumbersome, C code
and text file would be most efficient).
Dialplan
Exten => s,1,answer
Exten => s,n,AGI(check_bl.agi,${CALLERID(num)},${CALLERID(name)})
Exten => s,n,Gotoif($[${BLACKVAL} = "OK"]?6]
Exten => s,n,playback(nice-goodbye)
Exten => s,n,hangup
Exten => s,n,noop(caller is "ok", continue)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Myles Wakeham
Sent: Thursday, May 20, 2010 11:43 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Checking blank CallerID in Dialplan
I am trying to implement a change to our Dialplan that will thwart
tele-spammers that are calling us with blanked out caller ID.
The caller IDs seem to vary between originating callers when they block
caller ID. I've seen the following:
"anonymous"
""
So I'm checking for these. However recently one company seems to be
bypassing this, so what I wanted to do was implement some logic that
checks for actual numbers in the caller ID.
We have a couple of different SIP providers for incoming calls. Some
prefix numbers with a + and others don't. But I'm logging incoming
calls that are getting through our tele-spam filter and it seems that
they are blank, but I suspect they contain empty spaces which is why our
matches don't work.
Does anyone have some sample DialPlan code that they are using to thwart
incoming calls with no caller ID? I was thinking of maybe converting
the caller ID num to a numeric value and testing for 'not equal to 0'
but that won't work with the + prefix.
All suggestions greatly appreciated.
Myles
--
-
Myles Wakeham
Director of Engineering
Tech Solutions USA, Inc.
www.techsolusa.com
Phone +1-480-451-7440
--
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Re: [asterisk-users] Softphones on thin clients...
> -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes > Sent: Thursday, May 20, 2010 1:51 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Softphones on thin clients... > > > On 20 May 2010, at 18:35, Carlos Chavez wrote: > > I am worried about conflicts when running 10 softphones on the same > > server since they will all try to use por 5060. > > And the fact most terminal services servers/clients still don't support > audio input.. only output.. Since the little box has a MIC jack I suppose that it should support audio input. These boxes will be running Windows and using Eyebeam. -- 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] Softphones on thin clients...
Don't some thin clients run on WindowsCE or Linux/rdesktop? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Thursday, May 20, 2010 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Softphones on thin clients... On 20 May 2010, at 18:35, Carlos Chavez wrote: > I am worried about conflicts when running 10 softphones on the same > server since they will all try to use por 5060. And the fact most terminal services servers/clients still don't support audio input.. only output.. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphones on thin clients...
1. GPXE + HTTP 2. Tiny Core Linux 3. Profit... ~ Andrew "lathama" Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Thu, May 20, 2010 at 1:35 PM, Carlos Chavez wrote: > Does anyone know if you can use softphones on thin clients? I have a > new customer that wants to use Eyebeam (about 10 users) on a thin client > platform. Each user has a little box on their desk that has a USB port, > mic and headphone jacks and monitor. > > I am worried about conflicts when running 10 softphones on the same > server since they will all try to use por 5060. > > -- > Telecomunicaciones Abiertas de México S.A. de C.V. > Carlos Chávez Prats > Director de Tecnología > +52-55-91169161 ext 2001 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every 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] Attended Transfer using AMI
I am looking for a way to have an agent execute an attended transfer using the asterisk manager interface (AMI). I have been trying to use the dual Redirect from svn trunk. The problem with this function is that the "ExtraChannel" does not get redirected properly afaict. Now, I am looking for other solutions for the list, I will probably try playing DTMFs on the agent channel to simulate the manual transfer next unless anyone has some better ideas. Thanks Grant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Softphones on thin clients...
On 20 May 2010, at 18:35, Carlos Chavez wrote: > I am worried about conflicts when running 10 softphones on the same > server since they will all try to use por 5060. And the fact most terminal services servers/clients still don't support audio input.. only output.. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Softphones on thin clients...
Does anyone know if you can use softphones on thin clients? I have a new customer that wants to use Eyebeam (about 10 users) on a thin client platform. Each user has a little box on their desk that has a USB port, mic and headphone jacks and monitor. I am worried about conflicts when running 10 softphones on the same server since they will all try to use por 5060. -- 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] Cause and cure for "Exceptionally long voice queuelength queuing to Local"?
I don't see anything in the SIP trace related to the warning messages. Would anyone have any further tips? Thanks for any help! On Wed, May 19, 2010 at 9:12 PM, David Cunningham wrote: > What should I expect see if it is the peer asking us to slow down RTP? > > Thanks again. > > > On Wed, May 19, 2010 at 9:05 PM, Danny Nicholas wrote: >> Sip debug peer? >> >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David >> Cunningham >> Sent: Wednesday, May 19, 2010 3:00 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: [asterisk-users] Cause and cure for "Exceptionally long voice >> queuelength queuing to Local"? >> >> Hello, >> >> We're seeing lots of warnings like the following, running Asterisk >> 1.6.1.12. Does anyone know the cause or cure? >> >> One explanation I've come across is that the peer is congested and >> sending RTCP messages asking us to slow the RTP down. Is there any way >> we can verify this? >> >> [May 17 13:42:45] WARNING[27482] channel.c: Exceptionally long voice >> queue length queuing to Local/12126412...@asterisk-phone-7e3d;1 >> >> Thanks in advance! >> >> >> -- >> David Cunningham, Voisonics >> http://voisonics.com/ >> US toll-free: +1 888 842 2720 >> UK: +44 (0) 20 3298 1642 >> Australia: +61 (0) 2 9037 2180 >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every 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 >> > > > > -- > David Cunningham, Voisonics > http://voisonics.com/ > US toll-free: +1 888 842 2720 > UK: +44 (0) 20 3298 1642 > Australia: +61 (0) 2 9037 2180 > -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Checking blank CallerID in Dialplan
On May 20, 2010, at 12:43 PM, Myles Wakeham wrote: > I am trying to implement a change to our Dialplan that will thwart > tele-spammers that are calling us with blanked out caller ID. > > The caller IDs seem to vary between originating callers when they block > caller ID. I've seen the following: > > "anonymous" > "" > > So I'm checking for these. However recently one company seems to be > bypassing this, so what I wanted to do was implement some logic that > checks for actual numbers in the caller ID. > > We have a couple of different SIP providers for incoming calls. Some > prefix numbers with a + and others don't. But I'm logging incoming > calls that are getting through our tele-spam filter and it seems that > they are blank, but I suspect they contain empty spaces which is why our > matches don't work. > > Does anyone have some sample DialPlan code that they are using to thwart > incoming calls with no caller ID? I was thinking of maybe converting > the caller ID num to a numeric value and testing for 'not equal to 0' > but that won't work with the + prefix. > > All suggestions greatly appreciated. > > Myles In my opinion this is one of the greatest aspects of Asterisk... it's ability to do custom things just as this. Much like anything IT, there are many ways of approaching this. Different people have different solutions. One can be an API lookup to something such as whocalled.us (which has an AGI for asterisk). Another can be a localized black list. An yet another can be a localized white list. (If the number doesn't match a known, "approved" number, then give it some sort of IVR or screening) I don't get hit that much anymore, so I'm still with the modified blacklist approach. Unknown callers (00, 0123., Anonymous, Unknown, etc.) get screened and known telemarketers go to the torture I talked about at http://bit.ly/bbhho ---fred http://qxork.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] Checking blank CallerID in Dialplan
I am trying to implement a change to our Dialplan that will thwart tele-spammers that are calling us with blanked out caller ID. The caller IDs seem to vary between originating callers when they block caller ID. I've seen the following: "anonymous" "" So I'm checking for these. However recently one company seems to be bypassing this, so what I wanted to do was implement some logic that checks for actual numbers in the caller ID. We have a couple of different SIP providers for incoming calls. Some prefix numbers with a + and others don't. But I'm logging incoming calls that are getting through our tele-spam filter and it seems that they are blank, but I suspect they contain empty spaces which is why our matches don't work. Does anyone have some sample DialPlan code that they are using to thwart incoming calls with no caller ID? I was thinking of maybe converting the caller ID num to a numeric value and testing for 'not equal to 0' but that won't work with the + prefix. All suggestions greatly appreciated. Myles -- - Myles Wakeham Director of Engineering Tech Solutions USA, Inc. www.techsolusa.com Phone +1-480-451-7440 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Callerid with DAHDI
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Tzafrir. On Thu, May 20, 2010 at 09:58:26 -0300, Tzafrir Cohen wrote: >> I'm testing a telephone connected to FXS port of a Sangoma A200 card. >> But I'm observing that callerid is not working. The configuration >> that I'm using in chan_dahdi.conf is the following one: >> - - >> ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit >> ;autogenrated on 2010-05-11 >> ;Dahdi Channels Configurations >> ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak >> >> [trunkgroups] >> >> [channels] >> language=es >> defaultzone=es >> usecallerid=yes >> hidecallerid=no >> callwaiting=no >> threewaycalling=yes >> transfer=yes >> echocancel=yes >> echotraining=yes >> inmediate=no >> >> ; DGB - 20100322 >> busydetect=yes >> busycount=3 >> >> >> ;Sangoma AFT-A200 [slot:8 bus:1 span:1] >> context=from-internal >> group=1 >> echocancel=yes >> signalling = fxo_ks >> channel => 1 >> mailbox=...@voicemail >> callerid="Jane Doe" <300> > The 'mailbox' and 'callerid' settings only affect channels 2-4, and > not channel 1. Is that intentoinal? Hmmm ... then I just found out "mailbox" and "callerid" apply from where I put it and that its location is not trivial. Should these be over "channel"? Putting both under "context" I see that it works. Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkv1ZSYACgkQZpa/GxTmHTfHwACeKO9EHjtIoe+A5/UP+/KntPwg thoAn1EQMfSksCMgqSUUh63SNYlIjanX =wFrb -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Which issue is keeping you from updrading to 1.6.2 ?
On Thu, May 20, 2010 at 11:41 AM, Olivier wrote: > Hi, > > I'm evaluating what could keep me from upgrading production systems to > 1.6.2. > As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an > issue with BLF-pickup which kept me from going further. > > Have you met other issues I should include include in my checklist ? meetme CLI arguments changed between 1.6.0 and 1.6.2 Don't know where the delta was, and I haven't looked. I prefer the new syntax, and especially prefer the 'concise' option, but it might break features people have built in the past. Specifically, 1.6.0 'meetme' is replaced with 1.6.2 'meetme list' 1.6.2 'meetme' errors out, and requires an argument 1.6.0 'meetme list' would error out wanting a room argument 'meetme list ' works same on 1.6.0 and 1.6.2 'meetme list concise' and 'meetme list concise' in 1.6.2 are nice. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Which issue is keeping you from updrading to 1.6.2 ?
Hi, I'm evaluating what could keep me from upgrading production systems to 1.6.2. As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an issue with BLF-pickup which kept me from going further. Have you met other issues I should include include in my checklist ? 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] Adding a context from the console
Hi, this didn't seem to work. Is there something I am missing? dialplan add extension 1234,1,NoOp,hello into default Extension '1234,1,NoOp,hello' added into 'default' context -- Added extension '1234' priority 1 to default (0x8e8f520) dialplan add extension 1234,1,NoOp,hello into test Failed to add '1234,1,NoOp,hello' extension into 'test' context Regards Lee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Archer Sent: 19 May 2010 16:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Adding a context from the console Many thanks. Regards Lee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: 19 May 2010 16:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Adding a context from the console On Wednesday 19 May 2010 02:28:02 Lee Archer wrote: > Hi, is it possible to add a context from the console using the dialplan > command? Yes, just add an extension to it. The context will be created as needed. -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] OT - Which SIP hardphone with 100 BLF ?
On Thu, May 20, 2010 at 09:24:18AM -0400, SIP wrote: > Tzafrir Cohen wrote: > > On Wed, May 19, 2010 at 07:46:26AM +0200, Olivier wrote: > > > >> 2010/5/18 Danny Nicholas > >> > >> > >>> Dumb question – wouldn’t it be easier to monitor a web interface than a > >>> phone with 100 lights? > >>> > >>> > >> Yes and no : operator already has a Flash Operator Panel on its screen. > >> Information displayed by FOP is richer (you can see who is talking to who) > >> but operator feels easier with dedicated buttons for both displaying > >> activity and issuing transfers. > >> > >> I think 100 is the upper limit for both kinds of tools where at a glance, > >> you can see all extensions : I think above a certain user count (120 ?), > >> operator would prefer to specifically query its console to get current > >> specific extensions phone activity. > >> > > > > Just a thought: I have on my desktop a hardware device with some 100 or > > more buttons. No leds in them, sadly[1]. Remapping their labels is normally > > done using specialized hardware (sticky labels and the sort). > > > > Naturally there's the alternative of a touch screen. > > > > [1] A quick search found products such as > > http://blog.logitech.com/2009/10/15/new-logitech-gaming-keyboard-g110/ > > > > > Even IF you could get a keyboard with lights you could individually turn > on and off (never seen one), good luck getting a receptionist to use > it. I can picture it now... you hand your receptionist a lighted > keyboard and say 'make do,' and your receptionist brains you with said > keyboard when your back is next turned. Even IF you could get a phone with 100 buttons and such, good luck getting a receptionist to use it. I can picture it now... you hand your receptionist a phone with that lighted keyboard and say 'make do,' and your receptionist brains you with said keyboard when your back is next turned. > > There's a big difference between a workable situation and a complete and > utter kludge. The other difference is that it may force you to use an inferior (or way more expensive. Or both) phone for the receptionist. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ?
Your receptionist would wait until your back was turned? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of SIP Sent: Thursday, May 20, 2010 8:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT - Which SIP hardphone with 100 BLF ? Tzafrir Cohen wrote: > On Wed, May 19, 2010 at 07:46:26AM +0200, Olivier wrote: > >> 2010/5/18 Danny Nicholas >> >> >>> Dumb question - wouldn't it be easier to monitor a web interface than a >>> phone with 100 lights? >>> >>> >> Yes and no : operator already has a Flash Operator Panel on its screen. >> Information displayed by FOP is richer (you can see who is talking to who) >> but operator feels easier with dedicated buttons for both displaying >> activity and issuing transfers. >> >> I think 100 is the upper limit for both kinds of tools where at a glance, >> you can see all extensions : I think above a certain user count (120 ?), >> operator would prefer to specifically query its console to get current >> specific extensions phone activity. >> > > Just a thought: I have on my desktop a hardware device with some 100 or > more buttons. No leds in them, sadly[1]. Remapping their labels is normally > done using specialized hardware (sticky labels and the sort). > > Naturally there's the alternative of a touch screen. > > [1] A quick search found products such as > http://blog.logitech.com/2009/10/15/new-logitech-gaming-keyboard-g110/ > > Even IF you could get a keyboard with lights you could individually turn on and off (never seen one), good luck getting a receptionist to use it. I can picture it now... you hand your receptionist a lighted keyboard and say 'make do,' and your receptionist brains you with said keyboard when your back is next turned. There's a big difference between a workable situation and a complete and utter kludge. N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] file command with alaw file
Sox v14.1.0 doesn't play with alaw, but AFAIK, Asterisk has this function (this is from 1.4.30, think 1.6X has same functionality) CLI> help file convert Usage: file convert Convert from file_in to file_out. If an absolute path is not given, the default Asterisk sounds directory will be used. Example: file convert tt-weasels.gsm tt-weasels.ulaw -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pham Quy Sent: Thursday, May 20, 2010 2:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] file command with alaw file Hi, How can I convert FROM ALAW file, which generated by asterisk apps (monitor, or record app), to format (wav or mp3) that is playable by music player?? Can Sox do this? I have an asterisk 1.6.2.6 on my CentOS 5.2. I record audio clip by mixmonitor app and use file command to check the alaw output, and here is output: - 983006584-20100517-125002.alaw: RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz - How could file command recognize the format as there is no header in the output file? Or Did I probably miss something making asterisk yield incorrect alaw files? Please help, thanks Quyps On Thu, 2010-05-20 at 00:50 -0600, Steve Murphy wrote: > Quyps-- > > I've noticed in general that the ulaw, alaw, gsm, slin files used and > generated by > asterisk do not have headers (the RIFF stuff), and asterisk is not > expecting them. in general they > will louse up Asterisk's ability to play the sound. They are just raw > data and the extension > on the file name (.gsm, or .ulaw, etc) is the only clue to the file > format/contents. > > In general, if you need a sound file of your own making in a certain > format, you can convert > to most of the formats using sox in linux, but really, the best thing > to do is record the source > sound file in cd-quality sound WAV format, in 44 khz sampling rate, or > higher, and then > use sox to convert to 8khz format. Asterisk can do some of this via > the file convert CLI > command, ( on the asterisk cli, type "help file convert"). You'd have > to judge for yourself > if "file convert tt-weasels.gsm tt-weasels.ulaw" which would convert > the 8khz gsm format to > 8 khz ulaw, or "sox -v 0.7 tt-weasels.44khz.wav -r 8000 -c 1 -t sw > tt-weasels.raw;" > "sox -r 8000 -c 1 -t sw tt-weasels.raw -t ul tt-weasels.ulaw" which > is the way the Asterisk > sounds are produced from the the cd-quality sounds. They would seem a > bit equivalent. > > I wonder if just "sox -v 0.7 tt-weasels.44khz.wav -r 8000 -c 1 -t ul > tt-weasels.ulaw" would > sound any better... you audio engineers out there may have an opinion. > > I've personally noted that not all linux distributions provide the > same version of sox; > some distribute sox with an absolute minimum of sound formats > built-in. You may have > to go out and find all the libraries and roll your own sox. > > murf > > > > > > On Wed, May 19, 2010 at 10:34 PM, Pham Quy wrote: > On Mon, 2010-05-17 at 17:49 +0700, Pham Quy wrote: > > On Mon, 2010-05-17 at 13:06 +0700, Pham Quy wrote: > > > hi all, > > > > > > I install Asterisk 1.6 on Centos 5.2 (kernel 2.6.18-92.el5 > #1 SMP Tue > > > Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux) > and do record > > > audio clip with mixmonitor() as alaw file (softphone is > also configured > > > with alaw active only). Using file command i can get the > following > > > information > > > > > > 983006584-20100517-125002.alaw: RIFF (little-endian) data, > WAVE audio, > > > ITU G.711 A-law, mono 8000 Hz > > > > > > But when i install the same system on Centos 5.5 (kernel > 2.6.18-92.el5 > > > #1 SMP Tue Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 > GNU/Linux) i > > > could get the same information with file command. File > command > > > recognized alaw file as JPEG image: > > > > > > 983006584-20100517-123825.alaw: JPEG image data > > > > > > I guess i may miss something when i setup the new on on > Centos 5.5, but > > > u dont know what it is. Anyone have idea about this? > > > > > > please help. > > > > > > Thanks in advance. > > > Quyps > > > > I did check content of two alaw files (using a hex editor) > generated > > from two aboves cases. For the one fromo CentOS 5.2, > beginning bytes > > look likes : > > > > riff1^0.wavefmt@...@...fact.^0.data.^0... > > > > and the one from CentOS 5.5 > > > > ..RQVTVXEMBAX > > > > It seem like the first one have s
Re: [asterisk-users] run extensions after call moved to queue and answered by member
В Чтв, 20/05/2010 в 05:49 -0700, Jim Dickenson пишет: > Which version of asterisk are you running? Thank's for answer. One minute before i found answer - add membermacro to quesues.conf I'm use asterisk 1.6 -- Vasiliy G Tolstov Selfip.Ru -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] OT - Which SIP hardphone with 100 BLF ?
Tzafrir Cohen wrote: > On Wed, May 19, 2010 at 07:46:26AM +0200, Olivier wrote: > >> 2010/5/18 Danny Nicholas >> >> >>> Dumb question – wouldn’t it be easier to monitor a web interface than a >>> phone with 100 lights? >>> >>> >> Yes and no : operator already has a Flash Operator Panel on its screen. >> Information displayed by FOP is richer (you can see who is talking to who) >> but operator feels easier with dedicated buttons for both displaying >> activity and issuing transfers. >> >> I think 100 is the upper limit for both kinds of tools where at a glance, >> you can see all extensions : I think above a certain user count (120 ?), >> operator would prefer to specifically query its console to get current >> specific extensions phone activity. >> > > Just a thought: I have on my desktop a hardware device with some 100 or > more buttons. No leds in them, sadly[1]. Remapping their labels is normally > done using specialized hardware (sticky labels and the sort). > > Naturally there's the alternative of a touch screen. > > [1] A quick search found products such as > http://blog.logitech.com/2009/10/15/new-logitech-gaming-keyboard-g110/ > > Even IF you could get a keyboard with lights you could individually turn on and off (never seen one), good luck getting a receptionist to use it. I can picture it now... you hand your receptionist a lighted keyboard and say 'make do,' and your receptionist brains you with said keyboard when your back is next turned. There's a big difference between a workable situation and a complete and utter kludge. N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] OT - Which SIP hardphone with 100 BLF ?
On Wed, May 19, 2010 at 07:46:26AM +0200, Olivier wrote: > 2010/5/18 Danny Nicholas > > > Dumb question – wouldn’t it be easier to monitor a web interface than a > > phone with 100 lights? > > > Yes and no : operator already has a Flash Operator Panel on its screen. > Information displayed by FOP is richer (you can see who is talking to who) > but operator feels easier with dedicated buttons for both displaying > activity and issuing transfers. > > I think 100 is the upper limit for both kinds of tools where at a glance, > you can see all extensions : I think above a certain user count (120 ?), > operator would prefer to specifically query its console to get current > specific extensions phone activity. Just a thought: I have on my desktop a hardware device with some 100 or more buttons. No leds in them, sadly[1]. Remapping their labels is normally done using specialized hardware (sticky labels and the sort). Naturally there's the alternative of a touch screen. [1] A quick search found products such as http://blog.logitech.com/2009/10/15/new-logitech-gaming-keyboard-g110/ -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday @12 Noon and 1PM
This week on VUC: 12 Noon EDT: Office KONNECT - phones that can connect to asterisk or be used without a pbx 1 PM EDT: Dan York on his new book "7 Deadliest UC Attacks" and the usual segments of VoIP and Asterisk news, and the VUC 1 minute rant. Info: http://vuc.me Conference bridges are active from 11:45 AM EDT on Fridays sip:200...@login.zipdx.com (Thanks to ZipDX conference bridge) skype:vuc.me (using Skype for Asterisk via PhoneFromHere.com - thx to them and to Digium) irc: #vuc on Freenode.net or http://vuc.me/irc via web IRC client /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] Callerid with DAHDI
On Mon, May 17, 2010 at 10:26:18PM -0300, Daniel Bareiro wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Hi all! > > I'm testing a telephone connected to FXS port of a Sangoma A200 card. > But I'm observing that callerid is not working. The configuration that > I'm using in chan_dahdi.conf is the following one: > > - - > ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit > ;autogenrated on 2010-05-11 > ;Dahdi Channels Configurations > ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak > > [trunkgroups] > > [channels] > language=es > defaultzone=es > usecallerid=yes > hidecallerid=no > callwaiting=no > threewaycalling=yes > transfer=yes > echocancel=yes > echotraining=yes > inmediate=no > > ; DGB - 20100322 > busydetect=yes > busycount=3 > > > ;Sangoma AFT-A200 [slot:8 bus:1 span:1] > context=from-internal > group=1 > echocancel=yes > signalling = fxo_ks > channel => 1 > mailbox=...@voicemail > callerid="Jane Doe" <300> The 'mailbox' and 'callerid' settings only affect channels 2-4, and not channel 1. Is that intentoinal? > > context=from-internal > group=1 > echocancel=yes > signalling = fxo_ks > channel => 2 > > context=from-zaptel > group=0 > echocancel=yes > signalling = fxs_ks > channel => 3 > > context=from-zaptel > group=0 > echocancel=yes > signalling = fxs_ks > channel => 4 > - - > > I was comparing this configuration with which I have in my house with a > OpenVox card, where callerid works, and the unique difference that I > found is that I'm using fxo_ls. Can be it the cause of the problem? Hardly likely. But if that is what you suspect: sed -i -e 's/fxo_ks/fxo_ls/' /etc/asterisk/chan_dahdi.conf asterisk -rx 'dahdi restart' -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] run extensions after call moved to queue and answered by member
Which version of asterisk are you running? Older versions allowed for an AGI to be called when the queued call got connected with an agent. "core show application queue" Queue(queuename[|options[|URL][|announceoverride][|timeout][|AGI]) The optional AGI parameter will setup an AGI script to be executed on the calling party's channel once they are connected to a queue member. Newer versions allow for either an AGI or a macro. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On May 20, 2010, at 4:47 AM, Vasiliy G Tolstov wrote: > Hello. > > Can You provide example, how can i run specific extension after incoming > call going into queue and answered (but not hangup). > > (i want to use System(echo .) after member of specific queue > answered a call); > > Thank You. > -- > Vasiliy G Tolstov > Selfip.Ru > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk T.38 Gateway code testing
hi, i made page for Asterisk T.38 Gateway code testing http://www.voip-info.org/wiki/view/Asterisk+T.38+Gateway Final T.38 Gateway API for asterisk 1.8 will be posted by Kevin Fleming later BUT Asterisk 1.8 is too far and we need t.38 gw now if you would like help/test current code(last patch from https://issues.asterisk.org/view.php?id=13405), please contact me i have 2 public testing machines connected over E1 PLEASE do not post bug reports to https://issues.asterisk.org/view.php?id=13405 because this patch cannot be included in 1.6.2 (digium rules) i'm in contact with klaus darilion and daniel ferenci(asterisk t.38 developers) and i can arrange fixing bugs my jabber is cerv...@njs.netlab.cz look forward for better t.38 days --- Marek Cervenka jabber - cerv...@njs.netlab.cz === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] run extensions after call moved to queue and answered by member
Hello. Can You provide example, how can i run specific extension after incoming call going into queue and answered (but not hangup). (i want to use System(echo .) after member of specific queue answered a call); Thank You. -- Vasiliy G Tolstov Selfip.Ru -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] DAHDI and ESXi
The following link may be a suitable workaround How do I change the type of line from E1 to T1/J1 without using jumpers? http://kb.digium.com/entry/121/ Alec Davis -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Conor McTernan Sent: Thursday, 20 May 2010 7:09 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DAHDI and ESXi Does anybody have any experience of running Asterisk with DAHDI on ESXi? I am running Asterisk 1.4 with DAHDI 2.3 on ESXi 4.0 alongside a TE220 card. My asterisk install can see the card, but no matter what I do with the jumpers it remains in E1 mode. I have tested the card in another machine, in a non-virtualized setup and it works correctly (i.e. dahdi_scan reports the card as being in T1 mode). The issue appears to be with ESX, but I cannot understand why it would be ignoring (or even changing) the jumper settings. Has anyone else used any Digium cards in a virtualized environment. I am running all this on a Xeon E5506 so the PCI passthrough will work. Any ideas or comments are much appreciated. Thank you, Conor -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Sending fake auth rejection for user
It seems to be 401 unauthorized, your end point credentials are not correct On Thu, May 20, 2010 at 1:30 PM, Jonas Kellens wrote: > What does this mean : > > [May 20 09:57:28] NOTICE[14916]: chan_sip.c:15644 handle_request_subscribe: > Sending fake auth rejection for user ;tag=wetpp2qb3f > [May 20 09:57:28] NOTICE[14916]: chan_sip.c:15644 handle_request_subscribe: > Sending fake auth rejection for user ;tag=6pwd6erg54 > [May 20 09:57:29] NOTICE[14916]: chan_sip.c:15644 handle_request_subscribe: > Sending fake auth rejection for user ;tag=wetpp2qb3f > [May 20 09:57:29] NOTICE[14916]: chan_sip.c:15644 handle_request_subscribe: > Sending fake auth rejection for user ;tag=6pwd6erg54 > > > > Kind regards, > > Jonas. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Thank you with regards, Gopalakrishnan A.N, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Sending fake auth rejection for user
What does this mean : [May 20 09:57:28] NOTICE[14916]: chan_sip.c:15644 handle_request_subscribe: Sending fake auth rejection for user ;tag=wetpp2qb3f [May 20 09:57:28] NOTICE[14916]: chan_sip.c:15644 handle_request_subscribe: Sending fake auth rejection for user ;tag=6pwd6erg54 [May 20 09:57:29] NOTICE[14916]: chan_sip.c:15644 handle_request_subscribe: Sending fake auth rejection for user ;tag=wetpp2qb3f [May 20 09:57:29] NOTICE[14916]: chan_sip.c:15644 handle_request_subscribe: Sending fake auth rejection for user ;tag=6pwd6erg54 Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] file command with alaw file
Hi, How can I convert FROM ALAW file, which generated by asterisk apps (monitor, or record app), to format (wav or mp3) that is playable by music player?? Can Sox do this? I have an asterisk 1.6.2.6 on my CentOS 5.2. I record audio clip by mixmonitor app and use file command to check the alaw output, and here is output: - 983006584-20100517-125002.alaw: RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz - How could file command recognize the format as there is no header in the output file? Or Did I probably miss something making asterisk yield incorrect alaw files? Please help, thanks Quyps On Thu, 2010-05-20 at 00:50 -0600, Steve Murphy wrote: > Quyps-- > > I've noticed in general that the ulaw, alaw, gsm, slin files used and > generated by > asterisk do not have headers (the RIFF stuff), and asterisk is not > expecting them. in general they > will louse up Asterisk's ability to play the sound. They are just raw > data and the extension > on the file name (.gsm, or .ulaw, etc) is the only clue to the file > format/contents. > > In general, if you need a sound file of your own making in a certain > format, you can convert > to most of the formats using sox in linux, but really, the best thing > to do is record the source > sound file in cd-quality sound WAV format, in 44 khz sampling rate, or > higher, and then > use sox to convert to 8khz format. Asterisk can do some of this via > the file convert CLI > command, ( on the asterisk cli, type "help file convert"). You'd have > to judge for yourself > if "file convert tt-weasels.gsm tt-weasels.ulaw" which would convert > the 8khz gsm format to > 8 khz ulaw, or "sox -v 0.7 tt-weasels.44khz.wav -r 8000 -c 1 -t sw > tt-weasels.raw;" > "sox -r 8000 -c 1 -t sw tt-weasels.raw -t ul tt-weasels.ulaw" which > is the way the Asterisk > sounds are produced from the the cd-quality sounds. They would seem a > bit equivalent. > > I wonder if just "sox -v 0.7 tt-weasels.44khz.wav -r 8000 -c 1 -t ul > tt-weasels.ulaw" would > sound any better... you audio engineers out there may have an opinion. > > I've personally noted that not all linux distributions provide the > same version of sox; > some distribute sox with an absolute minimum of sound formats > built-in. You may have > to go out and find all the libraries and roll your own sox. > > murf > > > > > > On Wed, May 19, 2010 at 10:34 PM, Pham Quy wrote: > On Mon, 2010-05-17 at 17:49 +0700, Pham Quy wrote: > > On Mon, 2010-05-17 at 13:06 +0700, Pham Quy wrote: > > > hi all, > > > > > > I install Asterisk 1.6 on Centos 5.2 (kernel 2.6.18-92.el5 > #1 SMP Tue > > > Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux) > and do record > > > audio clip with mixmonitor() as alaw file (softphone is > also configured > > > with alaw active only). Using file command i can get the > following > > > information > > > > > > 983006584-20100517-125002.alaw: RIFF (little-endian) data, > WAVE audio, > > > ITU G.711 A-law, mono 8000 Hz > > > > > > But when i install the same system on Centos 5.5 (kernel > 2.6.18-92.el5 > > > #1 SMP Tue Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 > GNU/Linux) i > > > could get the same information with file command. File > command > > > recognized alaw file as JPEG image: > > > > > > 983006584-20100517-123825.alaw: JPEG image data > > > > > > I guess i may miss something when i setup the new on on > Centos 5.5, but > > > u dont know what it is. Anyone have idea about this? > > > > > > please help. > > > > > > Thanks in advance. > > > Quyps > > > > I did check content of two alaw files (using a hex editor) > generated > > from two aboves cases. For the one fromo CentOS 5.2, > beginning bytes > > look likes : > > > > riff1^0.wavefmt@...@...fact.^0.data.^0... > > > > and the one from CentOS 5.5 > > > > ..RQVTVXEMBAX > > > > It seem like the first one have some information about file > format, > > which make our convert tool works correctly, and which the > second one > > doesnt have. > > > > Can you explain to me this different, and how can i get the > information > > as the first one? > > > > Thanks in advances, > > Quyps > > This question have been asked for a while, I really need some > help > here? > > Thanks in advance. > Quyps > > > -- > _ > -- B
[asterisk-users] DAHDI and ESXi
Does anybody have any experience of running Asterisk with DAHDI on ESXi? I am running Asterisk 1.4 with DAHDI 2.3 on ESXi 4.0 alongside a TE220 card. My asterisk install can see the card, but no matter what I do with the jumpers it remains in E1 mode. I have tested the card in another machine, in a non-virtualized setup and it works correctly (i.e. dahdi_scan reports the card as being in T1 mode). The issue appears to be with ESX, but I cannot understand why it would be ignoring (or even changing) the jumper settings. Has anyone else used any Digium cards in a virtualized environment. I am running all this on a Xeon E5506 so the PCI passthrough will work. Any ideas or comments are much appreciated. Thank you, Conor -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Early injecting Jack between call parties
I use Jack for getting callee sound. Dial with option M(): JACK_HOOK(manipulate,i(rec_737219:input),o(rec_737219:output),c(rec_737219))=on This works fine, but I need to connect the sound channel to Jack *before* the actual answer. As you can see in the AMI log, between "Ringing" to JACK_HOOK there is a 6 second break. I don't want that. I need a way to launch Dialplan function right after the channel is open. Is it safe to launch JACK_HOOK to a channel that just started to ring? (from AMI)? I doing it from the dialplan, because there is no interaction (yet) from the AMI with our asterisk. Version 1.6.2.7 human_now: 2010-05-20 01:42:03.567385 Event: Newexten Privilege: dialplan,all Timestamp: 1274308923.567385 Channel: SIP/Prov6-01be Context: NPDB2 Extension: 3706264 Priority: 75 Application: Dial AppData: SIP/GW1/0073706264,60,M(connect-jack,737219) Uniqueid: 1274308923.446 human_now: 2010-05-20 01:42:03.568501 Event: Dial Privilege: call,all Timestamp: 1274308923.568501 SubEvent: Begin Channel: SIP/Prov6-01be Destination: SIP/GW1-01bf CallerIDNum: CallerIDName: UniqueID: 1274308923.446 DestUniqueID: 1274308923.447 Dialstring: GW1/0073706264 human_now: 2010-05-20 01:42:10.646068 Event: Newstate Privilege: call,all Timestamp: 1274308930.646068 Channel: SIP/GW1-01bf ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 3706264 CallerIDName: Uniqueid: 1274308923.447 human_now: 2010-05-20 01:42:16.905284 Event: Newstate Privilege: call,all Timestamp: 1274308936.905284 Channel: SIP/GW1-01bf ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 3706264 CallerIDName: Uniqueid: 1274308923.447 human_now: 2010-05-20 01:42:16.905449 Event: Newexten Privilege: dialplan,all Timestamp: 1274308936.905449 Channel: SIP/GW1-01bf Context: macro-connect-jack Extension: s Priority: 1 Application: NoOp AppData: SIP/GW1-01bf Uniqueid: 1274308923.447 human_now: 2010-05-20 01:42:16.905470 Event: Newexten Privilege: dialplan,all Timestamp: 1274308936.905470 Channel: SIP/GW1-01bf Context: macro-connect-jack Extension: s Priority: 2 Application: Set AppData: JACK_HOOK(manipulate,i(rec_737219:input),o(rec_737219:output),c(rec_737219))=on Uniqueid: 1274308923.447 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] file command with alaw file
Quyps-- I've noticed in general that the ulaw, alaw, gsm, slin files used and generated by asterisk do not have headers (the RIFF stuff), and asterisk is not expecting them. in general they will louse up Asterisk's ability to play the sound. They are just raw data and the extension on the file name (.gsm, or .ulaw, etc) is the only clue to the file format/contents. In general, if you need a sound file of your own making in a certain format, you can convert to most of the formats using sox in linux, but really, the best thing to do is record the source sound file in cd-quality sound WAV format, in 44 khz sampling rate, or higher, and then use sox to convert to 8khz format. Asterisk can do some of this via the file convert CLI command, ( on the asterisk cli, type "help file convert"). You'd have to judge for yourself if "file convert tt-weasels.gsm tt-weasels.ulaw" which would convert the 8khz gsm format to 8 khz ulaw, or "sox -v 0.7 tt-weasels.44khz.wav -r 8000 -c 1 -t sw tt-weasels.raw;" "sox -r 8000 -c 1 -t sw tt-weasels.raw -t ul tt-weasels.ulaw" which is the way the Asterisk sounds are produced from the the cd-quality sounds. They would seem a bit equivalent. I wonder if just "sox -v 0.7 tt-weasels.44khz.wav -r 8000 -c 1 -t ul tt-weasels.ulaw" would sound any better... you audio engineers out there may have an opinion. I've personally noted that not all linux distributions provide the same version of sox; some distribute sox with an absolute minimum of sound formats built-in. You may have to go out and find all the libraries and roll your own sox. murf On Wed, May 19, 2010 at 10:34 PM, Pham Quy wrote: > On Mon, 2010-05-17 at 17:49 +0700, Pham Quy wrote: > > On Mon, 2010-05-17 at 13:06 +0700, Pham Quy wrote: > > > hi all, > > > > > > I install Asterisk 1.6 on Centos 5.2 (kernel 2.6.18-92.el5 #1 SMP Tue > > > Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux) and do record > > > audio clip with mixmonitor() as alaw file (softphone is also configured > > > with alaw active only). Using file command i can get the following > > > information > > > > > > 983006584-20100517-125002.alaw: RIFF (little-endian) data, WAVE audio, > > > ITU G.711 A-law, mono 8000 Hz > > > > > > But when i install the same system on Centos 5.5 (kernel 2.6.18-92.el5 > > > #1 SMP Tue Jun 10 18:51:06 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux) i > > > could get the same information with file command. File command > > > recognized alaw file as JPEG image: > > > > > > 983006584-20100517-123825.alaw: JPEG image data > > > > > > I guess i may miss something when i setup the new on on Centos 5.5, but > > > u dont know what it is. Anyone have idea about this? > > > > > > please help. > > > > > > Thanks in advance. > > > Quyps > > > > I did check content of two alaw files (using a hex editor) generated > > from two aboves cases. For the one fromo CentOS 5.2, beginning bytes > > look likes : > > > > riff1^0.wavefmt@...@...fact.^0.data.^0... > > > > and the one from CentOS 5.5 > > > > ..RQVTVXEMBAX > > > > It seem like the first one have some information about file format, > > which make our convert tool works correctly, and which the second one > > doesnt have. > > > > Can you explain to me this different, and how can i get the information > > as the first one? > > > > Thanks in advances, > > Quyps > > This question have been asked for a while, I really need some help > here? > > Thanks in advance. > Quyps > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Steve Murphy ParseTree Corp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
