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../libtool: line 1231: cygpath: command not found You need to put cygpath in your PATH. This might also be why configure is failing. Best, Tristan --20cf307f35267e30eb0505f59648 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: quoted-printable <div dir=3D"ltr">Hi,<br><div class=3D"gmail_extra"><br><div class=3D"gmail_=

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

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. > >

... 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

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

<div style="FONT-FAMILY: Arial; COLOR: rgb(0, 0, 0); FONT-SIZE: 12px"><div>Thanks all for the suggested tips. I confess I tried VMWare hypervisor esxi and found it less complicated to get set up and functioning correctly. <br /><br />I'll have to take up KVM another day, when I'm in less of a hurry to get something up and running right away. <br

2008-01-18 22:04 +0000 [r99080-99085] Russell Bryant <russell at digium.com> * CREDITS, include/asterisk/http.h, main/tcptls.c (added), main/manager.c, channels/chan_sip.c, doc/siptls.txt (added), main/Makefile, main/http.c, include/asterisk/tcptls.h (added), configs/sip.conf.sample, CHANGES: Merge changes from team/group/sip-tcptls This set of changes

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

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

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

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

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

described (stop accepting calls and shut down when all calls have completed). If you don't want to stop accepting calls, but still want to stop Asterisk when there are no active calls, you can use "stop when convenient". The same qualifiers ("gracefully" and "when convenient") can be applied to the "restart" command. Cheers, AR On Dec 10, 2007 7:29 AM,

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

... 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

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: >

<div dir="ltr"><br> </div><div dir="ltr">Actually I want to run some analysis pass only on the user defined functions but not on the library functions. Is there any boolean method that can tell which is a library function and which is not? </div><div class="wps_quotion">On 5 Feb 2020 5:23 a.m., David Blaikie

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

user, password from user_sensitive_data_table into dovecot-sql.conf, but I'll live with that. You most probably had your reasons, and ultimately I agree - security first ;-) -- Chaos greets U ------=_Part_57551_1009602.1160777305352 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: 7bit Content-Disposition: inline <br><br><div><span

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type"> </head> <body bgcolor="#ffffff" text="#000000"> Talin wrote: <blockquote