similar to: No subject

Displaying 20 results from an estimated 5000 matches similar to: "No subject"

2009 Jan 16
0
No subject
... 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
2007 Jul 12
0
No subject
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
2007 Aug 16
0
No subject
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
2007 Jul 12
0
No subject
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
2007 Jul 12
0
No subject
... 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
2007 Jul 12
0
No subject
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
2007 Jul 12
0
No subject
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
2008 Mar 25
0
No subject
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
2007 Jul 12
0
No subject
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
2011 Sep 02
0
No subject
penSuse 12.1. Lets check with OpenSuse 12.1.&nbsp; <div><br /> </div> <div>Regards.</div> <div><br /> <br /> <div class=3D"gmail_quote">On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan = N <span dir=3D"ltr">&lt;<a href=3D"mailto:gopalakrishnan.an at gmail.com" targ=
2009 Jan 16
0
No subject
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
2007 Jul 12
0
No subject
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,
2007 Jul 12
0
No subject
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
2011 Jan 06
0
No subject
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
2013 Apr 11
0
No subject
../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_=
2007 Jul 12
0
No subject
you think ? > > ** Login/Logout of queues, Day/Night mode buttons with indication (1.6 > has this as well). > ** Company internal directory on the phone updated on the PBX Some (most ?) IP phones support this > > ** System Speed Dial on the display updated by the PBX This one is interesting. I can't see a way to do it. Ant idea ? > > ** Call Fwd by PBX with LED
2011 Apr 12
0
No subject
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
2009 Jul 20
0
No subject
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: >
2011 May 13
0
[LLVMdev] [ptx] Propose a register class naming convention change
<!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"
2007 Jul 12
0
No subject
Leg/Transaction Does Not Exist" and obviously not taken into account as endpoint GUI remains unchanged. Looking deeper into this here are : NOTIFY message accepted by S450IP NOTIFY sip:7531 at 192.168.100.197:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK06adc48b;rport From: "asterisk" <sip:asterisk at asterisk>;tag=as4ea953db To: <sip:sip:7531 at