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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

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

sses, that way autoloading works ok and the classes are found, but that see= ms a bit awkward. <br></div><blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid= rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br><br= >Note that it&#39;s a bit redundant to name your classes that way -- you<br= > can just as

tnet.itand SIP register messages are not replied. I suggested to check if your Asterisk box is really sending SIP messages, you can use a net sniffer. Did you alerady used different sip client with the same sip account of your Asterisk box? Did you use zoiper from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo < gincantalupo at

Olle ?) aiming to unify logging, eventing, monitoring (AMI, SNMP, ...) APIs. I think that thread occurred when it was decided to include a version number in Manager interface. I agree this is an interesting idea ... The use case that made me ask this is here : I've got a running system which is working ok up to a moment it stops to dial out on ISDN-BRI spans (incoming calls are ok). When

... Activating "sip debug" shows the register packets but nothing in return. ... I think that this is a network related issue, but you have to solve it by using a Asterisk config file. Unfortunately I think that the faster way to solve your problem is trying to understand if sip messages are correctly sent to tnet. I strongly suggest to use http://www.wireshark.org/ previoulsly named

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

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

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: >

<!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"

1. You pass in half the samples as the 'bits' arg. Speex looks at 1 frame worth of those bits and decodes them, decoded result in 'pcm'. 2. You pass in exactly 1 frame of data as the 'bits' arg. Speex looks at 1 frame worth of those bits (which is all there, exactly), decodes them, stores decoded result in 'pcm'. 3. You pass in 2 frames of data as

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 >

Hi, As you may know, these ISDN BRI features are very important here in Europe as ISDN Basic Rate Access is very popular among Small & Medium Entreprises. I don't really know why but it seems that in many countries, default is to install small PBX using Point-to-Multipoint (PtMP) mode as opposed to Point-to-Point (PtP) which is the norm for PRI. So basically, in several countries, SME

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