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supported, beside Idle, On call and Ringing ? Can we expect this list to match DEVICE_STATE's one (UNKNOWN | NOT_INUSE | INUSE | BUSY | INVALID | UNAVAILABLE | RINGING | RINGINUSE | ONHOLD) > Might be worth seeing if other phones do the same. > > S > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by

Let's say I have two Asterisk boxes, A and B. I am trying to get A to do SIP registration on B, so an extension for A can dial SIP phones covered by B. If I examine the logs on B, if the registration succeeds, I am seeing a notice to that effect on B. But if the registration *fails*, i'm not seeing any message logged on B. Maybe I'm just not looking in the right place. Is there a

mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? > > However, the MWI does not indicate voice mails for 610 and I keep seeing > this error message: > > ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox > 610 in context a10 > > However, mailbox 610 is clearly defined in voicemail.conf: >

have adaptors compatible with Asterisk, but explicitly say in the product titles that they're unlocked, which I think is the key. On Thu, Dec 17, 2009 at 4:16 AM, Brian Cline <Brian at nw.brian.fm> wrote: > Hello, > > I'm running Asterisk v1.6.1.11 internally with a few Linksys SIP > phones and will be receiving a machine containing a Dialogic card > for a

an external program, which at this stage, is not customizable ... I don't know if alternatives (LiMO, Android, ...) would be more open to this customization but for Symbian, not only Nokia SIP client wouldn't let you autoanswer to SIP calls, but any other SIP client complying to Symbian design wouldn't support autoanswer. PS: Please, note that I'm far from being an expert in GSM

If you don't use 'CERTVERIFY 1', then this will at least make sure that nobody can sniff your sessions without a large effort (...) > So, do I misunderstand CERTVERIFY directive ? Or is there a bug ? >> Can you reproduce such behaviour ? >> > > I'm not sure what is going on. Can you try running 'upsmon' with debugging > enabled? The following are

That's the main reason I opened this thread as it surprised me a bit ... > > > Any 2-wire analog leg will be a source of echo. Many, many, many calls > do not have a 2-wire leg. Even in handset audio circuit ? I was thinking that any handset is a potential echo source due to this audio circuit ... Do you agree ? > Think cell/mobile or endpoints with PRI or T-1. > >

such file or directory" on pure-IP platform in which I installed asterisk-libpri-dahdi trilogy. Maybe, it's me while following README instructions, maybe README instructions could be improved or maybe it's wrongly labeled messages ? That's why I told myself : I'm waiting too much from doc ? is a pure-IP platform too specific ? what is the official policy ? README starts with

... Thanks, anyway for telling as at least, it reflects your needs. > > > You want NT PtMP and i second that, > not being limited on the asterisk > side is a must in the > telephony ecosystem, since the legacy PABX aren't alwsys easy to > reconfigure. > > _______________________________________________ > -- Bandwidth and Colocation Provided by

connecting legacy PBX to Asterisk (for the very same reason, those PBX use TE-PTMP). If others could join this thread and say if they agree or not with NT-PTMP being the 2nd most needed mode, would be great. Please, do not hesitate to comment. > > > Right now, I would not preclude the possibility that NT-PTMP support > might be added, but I could not give you a concrete time at which

could be "hot". Is there any chance this would cause the card to fail after a while? It appears this site just had 4 port Digium card fail today. > Also, I am trying to cross connect with another Asterisk system with > > the normal LBO setting (i.e. span=1,1,0,esf,b8zs) but as of yet the > > systems aren't seeing each other at all. Could the side with the high >

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=UTF-8" http-equiv="Content-Type"> </head> <body bgcolor="#ffffff" text="#000000"> Justin Holewinski wrote: <blockquote cite="mid:BANLkTi=Y9EFmWRu-9dQxydq8zTyF7tEbJw@mail.gmail.com"

at least once a week I receive such an attack coming from a different ip. I will read the articles. Thanks again to everyone. Regards, Rodrigo Lang. 2010/6/29 Kenny Watson <kwatson at geniusgroupltd.com> > Hi, you can use fail2ban >

1. Is it normal to see : # lsmod Module Size Used by dahdi_dummy 3236 0 Shouldn't it be used by asterisk or is this 0 value meaning something specific ? 2. How can you check dahdi is running ? Here, "ps aux | grep dahdi " replies "grep dahdi". Cheers ------=_Part_2692_19661943.1228286635399 Content-Type: text/html; charset=ISO-8859-1

I'm not aware of any zaptel driver for such HFC USB modem (some Xorcom's products use USB, so ...) so I'm inclined to think it's not possible but it's better to ask ... Cheers ------=_Part_13548_28463665.1207585749504 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: quoted-printable Content-Disposition: inline Hi

"Why Siphon doesn't allow to receive a call when it doesn't run Apple doesn't accept (for the moment) an application runs in the background= . So, when Siphon doesn't run, the SIP server of your provider doesn't know your iPhone." --0015174c3c60a73ef5046656ca27 Content-Type: text/html; charset=windows-1252 Content-Transfer-Encoding: quoted-printable

takes precedence over a queue's defined moh class. --=20 Thanks, --Warren Selby, dCAP http://www.selbytech.com --000e0ce0494051d402049b4247c1 Content-Type: text/html; charset=windows-1252 Content-Transfer-Encoding: quoted-printable <div class=3D"gmail_quote">On Tue, Feb 1, 2011 at 10:20 AM, Danny Nicholas = <span dir=3D"ltr">&lt;<a

sort of standard for getting media players to support dynamically mixing different tracks and also making it easy for artists to do. On Mon, Aug 18, 2008 at 7:09 PM, Andy <andycool22 at gmail.com> wrote: > i'll chime in and say that i would love to get music recorded in > separate tracks, maybe there would be some kind of settings embedded > in the files so i could hear them

sort of standard for getting media players to support dynamically mixing different tracks and also making it easy for artists to do. On Mon, Aug 18, 2008 at 7:09 PM, Andy <andycool22 at gmail.com> wrote: > i'll chime in and say that i would love to get music recorded in > separate tracks, maybe there would be some kind of settings embedded > in the files so i could hear them

sort of standard for getting media players to support dynamically mixing different tracks and also making it easy for artists to do. On Mon, Aug 18, 2008 at 7:09 PM, Andy <andycool22 at gmail.com> wrote: > i'll chime in and say that i would love to get music recorded in > separate tracks, maybe there would be some kind of settings embedded > in the files so i could hear them