Ishanka Anuradha Ranasooriya
2006-Nov-29 20:15 UTC
[asterisk-users] Re: asterisk-users Digest, Vol 28, Issue 152
asterisk-users-request@lists.digium.com wrote:> Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > > You can reach the person managing the list at > asterisk-users-owner@lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > > 1. beeping noise in background (Kim Jones) > 2. RE: What's up with the Manager Interface?!?! (Douglas Garstang) > 3. g726 voice prompts (Eric Bishop) > 4. Re: What's up with the Manager Interface?!?! (Tony Mountifield) > 5. RE: What's up with the Manager Interface?!?! (Douglas Garstang) > 6. Cisco 7940 Firmware 8.2 (James R. Stevens) > 7. Re: voicemail.conf locking problem (Michiel van Baak) > 8. Re: What's up with the Manager Interface?!?! (Richard Lyman) > 9. Call recording with Asterisk BE (Ed Nu?ez) > 10. Re: voicemail.conf locking problem (Tzafrir Cohen) > 11. Re: How to park calls on a specific extension (Steve Sobol) > 12. RE: What's up with the Manager Interface?!?! (Douglas Garstang) > 13. Call dropping (Ed Nu?ez) > 14. Re: How to park calls on a specific extension (Steve Sobol) > 15. Re: SIP Port 5060 (Joseph) > 16. RE: What's up with the Manager Interface?!?! (Douglas Garstang) > 17. Setting RTP ports for Asterisk? (Vincent Delporte) > 18. Re: Re: What's up with the Manager Interface?!?! (Richard Lyman) > 19. Re: What's up with the Manager Interface?!?! (Richard Lyman) > 20. Re: How to park calls on a specific extension (Ira) > 21. Re: SIP Port 5060 (Andrew Joakimsen) > 22. Re: Siemens Gigaset C450 IP vs S450 IP (Andrew Joakimsen) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 29 Nov 2006 16:06:57 -0600 > From: "Kim Jones" <kim@anmsinc.com> > Subject: [asterisk-users] beeping noise in background > To: <asterisk-users@lists.digium.com> > Message-ID: <000401c71402$aff83d40$052110ac@ANMS.pvt> > Content-Type: text/plain; charset="us-ascii" > > I have asterisk 1.2.12.1 running with several client phone options. Our > echo cancellation is finally working great. The only problem I seem to > be having is there is background noise including beeping sounds at > regular intervals no matter which phone we use. Does anyone know why? > We are using a diqium tdm card. > > Thanks > > Kim > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061129/81c3ce79/attachment-0001.htm > > ------------------------------ > > Message: 2 > Date: Wed, 29 Nov 2006 15:16:45 -0700 > From: "Douglas Garstang" <dgarstang@oneeighty.com> > Subject: RE: [asterisk-users] What's up with the Manager Interface?!?! > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <645FEC31A18FE54A8721500CDD55A7B6035D0C0D@mail.oneeighty.com> > Content-Type: text/plain; charset="iso-8859-1" > > >> -----Original Message----- >> From: Steve Edwards [mailto:asterisk.org@sedwards.com] >> Sent: Wednesday, November 29, 2006 2:55 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: RE: [asterisk-users] What's up with the Manager Interface?!?! >> >> >> On Wed, 29 Nov 2006, Douglas Garstang wrote: >> >> >>> Grrrr. Here's another example... >>> >>> Action: Command >>> Command: sip show peer 2944093 >>> >>> Response: Follows >>> Privilege: Command >>> >>> >>> * Name : 2944093 >>> Secret : <Set> >>> MD5Secret : <Not set> >>> Context : 180o_CallStart >>> Subscr.Cont. : 180o_WatchBLF >>> >>> Why the HELL is there an asterisk before 'Name'? Now I have >>> >> to strip the bloody thing out! >> >>> And why is there TWO empty lines before it? >>> Good grief! >>> >>> Doug. >>> >> Would it be a better use of your time to "fix" the offending modules >> rather than kludge your code to handle the inconsistencies? >> >> Is AMI spec'd or would that be the first step? >> > > Steve, > > No... I'm not a C programmer. A standard interface would be a first step. :) > > Doug. > > > ------------------------------ > > Message: 3 > Date: Thu, 30 Nov 2006 09:19:13 +1100 > From: "Eric Bishop" <asterisk.eric@gmail.com> > Subject: [asterisk-users] g726 voice prompts > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <4acda1b40611291419r3eb8651en34e5e0c7570f1fb2@mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Anyone know if it posible to make voice promps native g726 or g711 format? > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061129/4e4f4dfb/attachment-0001.htm > > ------------------------------ > > Message: 4 > Date: Wed, 29 Nov 2006 22:19:16 +0000 (UTC) > From: tony@softins.clara.co.uk (Tony Mountifield) > Subject: [asterisk-users] Re: What's up with the Manager Interface?!?! > To: asterisk-users@lists.digium.com > Message-ID: <ekl114$94n$1@softins.clara.co.uk> > > In article <456DDAE1.4050705@dynx.net>, > Richard Lyman <pchammer@dynx.net> wrote: > >> just wait till you get a 'hiccup' that causes a line to get cut off, >> drop a char, and continue on next line. <G> >> (examples below) >> > > I've made heavy use of the Manager interface for over 2 years now, and > have never seen the kind of behaviour you described and showed examples > of. I would be more inclined to suspect the functions you are using to > read and collect the AMI output. Perhaps there's a buffer boundary > error or something. > > Cheers > Tony >I have asterisk 1.2.13 running with several client phone options. I have a problem in configuring in asterisk. I configure asterisk meetme.conf and extension.conf, but when i transfer call to conference it give me this message and asterisk kill it self. Ouch ... error while writing audio data: : Broken pipe If any one knows about this please help me to fix this. Cheers Ishanka -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061129/b199421b/attachment.htm
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