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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

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

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. > >

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,

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

... 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

else. Stuart =20 ----- Original Message -----=20 From: Ivailo Karamanolev=20 To: flac-dev at xiph.org=20 Sent: Wednesday, January 06, 2010 7:49 AM Subject: Re: [Flac-dev] FLAC C API / Visual Studio 2008 FILE* Issue I thought about this, and the MSVCRT mismatch also. What annoys me is = that I even tried compiling the library myself (with exactly the same = Visual Studio 2008 as my

This is a multipart MIME message. --= Multipart Boundary 1009021339 Content-Type: text/plain; charset="ISO-8859-1" Content-Transfer-Encoding: 8bit Hello there Did you know that you can program smart cards with files from the internet and open lots of pay per view chanells for your televisual pleasure. Take a look at http://MagicFun.da.ru for the latest hex files. Many thanks Jay.

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

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

All; I have a Fedora 26 server that I am trying to compile asterisk-certified-13.13-cert6 on. However, I'm getting the following errors. I'm also having a tough time trying to compile Dahdi. I'm not sure what I'm missing, but if anyone else is running Fedora, I'd really appreciate any help at all. Thanks Much; John V. make[1]: Leaving directory

Would it be sufficiant if I were to copy the makefile and res_config_ldap.c to the res/ directory of my running Asterisk and do make; make install? or do I have to do LIBS=-lldap export LIBS ./configure before that? My asterisk version is 1.2.6. Thanks in advance, Abhishek * * * * On 8/27/07, Gavin Henry <gavin.henry at gmail.com> wrote: > > I see it is res_config_ldap. You'd be

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

Hi, I'm trying to connect to the asterisk pbx via wss, from sipml5.org demo page (http://sipml5.org/call.htm). I used the guide from https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial , to setup the tls. I could make a secure sip call ( SRTP) using the PhonerLite sip client. ( This confirms my sip - tls settings and tls certficates. ( I'd added the tls client certficate

Class: default File: /var/lib/asterisk/moh//reno_project-system File: /var/lib/asterisk/moh//macroform-robot_dity File: /var/lib/asterisk/moh//manolo_camp-morning_coffee File: /var/lib/asterisk/moh//macroform-cold_day File: /var/lib/asterisk/moh//macroform-the_simplicity Class: none File: /var/lib/asterisk/moh/.nomusic_reserved/silence

Trying to setup UM with Office 365 which requires TLS. I've tried under 1.8.5.0 and under 1.6.2.16.1 and I get the same error: [Aug 11 06:50:20] VERBOSE[3023] tcptls.c: SSL certificate ok [Aug 11 06:50:20] VERBOSE[3023] tcptls.c:?? == Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) [Aug 11 06:50:20] WARNING[3023] tcptls.c: FILE * open failed! Following the

Hi, I?m trying to get TLS to work with asterisk and client phones, and all I?m getting from asterisk is [Aug 23 11:46:42] WARNING[1170]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! == Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) [Aug 23 11:46:44] WARNING[1171]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! when clients try to

Yea, I have faked that with a silent mp3, but to do it right it should also be a config flag in the agent.conf file for each agent, prolly add another arg to each agent definition for the MOH class, & the arg 'none' means don't play music for that agent -----Original Message----- From: James Golovich <james@wwnet.net> To: asterisk-users@lists.digium.com

Hey all, I'm currently running Asterisk 1.8.3 with FreePBX 2.8.1.3. It's tied to another IP-PBX via TLS. I have two problems going on.. 1.) Every so often (say roughly every 24 hours), Asterisk stops handing calls back to the second IP-PBX. The call rings indefinitely and Asterisk complains about the certificate like below: a. [Mar 16 16:10:04] VERBOSE[2973] tcptls.c: SSL certificate ok