similar to: Sipura-3000 IP->PSTN scenrio

This post is simply documenting a spouse-friendly way of using the spa-3000 as both a fxs and fxo port for basic soho environments in the US, allowing asterisk to participate as needed/wanted. All home phones are connected _only_ to the spa-3000 fxs port. The incoming home pstn line is connected _only_ to the spa-3000 fxo port. Defined Line 1 (fxs) to register with asterisk via sip (extn

Hi everyone. I'm wondering whether a setup like the one described below would be possible with Tinc: nodeA1 nodeB3 | | (Network A) --- [GWA] --- (Internet) --- [GWB] --- (Network B) | | nodeA2

Yo all, As there has been some intrest, here's my updated version: I post it to "-dev" as well as "-users", as it may be of intrest to both. Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate set of features. Currently, my implementation supports call- forward unconditional, on no answer

> gw0#cop flash:1:c820-k9osy6-mz.122-15.T5.bin tftp > Address or name of remote host []? 10.100.0.1 > Destination filename [c820-k9osy6-mz.122-15.T5.bin]? > TFTP: error code 2 received - Only absolute filenames allowed tftp-hpa 0.34 handles relative paths fine here. Which version? does not look like any error codes I have ever seen. > %Error opening

Sorry to post this here, but since last week I've posted to Fedora's mailing list, pptp-devel's list, and linuxgeneral forum. No one has given much for answers, or the people that have replied have not understood what I'm looking for. Hoping someone here can. For the record, I'm doing this on FC3 not CentOS. Anyway, if anyone can provide assistance I would be very appreciative.

Hey ppl, This is probably something very basic i'm doing wrong but I can't seem to get aorund it after reading the man page, it goes between error 2 and 1, im trying to tftp from a cisco router to my linux box. my in.tftpd.conf file looks like this: INTFTPD_PATH="/usr/tftp" INTFTPD_OPTS="-l ${INTFTPD_PATH}" and ive done a chmod 777 on the above path just to be 100%

Hello, I would like to put 1 asterisk box in Country A and 1 asterisk box in Country B. Let's assume : - Asterisk box in country A = GWA - Asterisk box in country B = GWB - Calling party number (located in country A) = CgPNA - Called party number (located in country B) = CdPNB - Second Called party number (located in country B) = sCdPNB - PSTN in country A = PSTNA - PSTN in country B = PSTNB

I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some settings by combining the information from various places around the 'net. I have typed out

I'd like to announce the release of an open source connector bridge for Asterisk and UniMRCP. The connector bridge is an implementation of Asterisk's Generic Speech API using UniMRCP client stack. This module allows Asterisk to connect to MRCPv2 or MRCPv1 compliant servers for speech recognition. It also allows to offload Asterisk using client/server architecture MRCP provides. Moreover,

Have you thought about making them agents, they would both be reachable by dialing there agent number then, and I know only one agent can be logged in at once. Just a thought. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Saturday, April 22, 2006 8:26 PM To:

hello my friend has the proxy set up his extention set up his password set up but he isn't getting a dial tone is there a second setting we need to put the address in? he is going to advenced settings line1 and in the proxy address box he is putting the info in below is the way he has it set up Sipura SPA Configuration Sipura Technology Inc Info System SIP Provisioning Regional Line 1 User 1

Hi. I'm using Samba on FreeBSD to host various file servers. Recently I've noticed one weird thing: samba needs w flag on file for being able to delete it, plus, when 'force user' is used, samba needs additional flags for group owning the file. Not sure if it was there all the time and it's just me, or may be this is an intended behaviour (I hope not), so, anyway I'll

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

+ Adrian + David Hi Arsen, This sounds like a bug to me. Have you tried reproducing it on trunk? For instance, I see that the relation between DICompileUnit and DISubprogram was changed in the meantime (https://reviews.llvm.org/D19034 <https://reviews.llvm.org/D19034>). If this no longer occurs on master you could bisect the compiler to find the commit(s) that fix this and consider

Hello list, After upgrade to dovecot version 1.2.6 I have noticed some quota malfunctions(users become often overquota). Looking at logs closely I have found that quota value resets to general dovecot-config( quota = maildir:Mailbox quota quota_rule = *:storage=500M quota_rule2 = Trash:storage=10%% ), after dovecot --exec-mail ext /usr/libexec/dovecot/expire-tool.sh command. The problem that

Thanks Paul, Hi Adrian and David I would really appreciate any comments, thoughts assumptions. If additional information is needed please let me know. Regards, Arsen On Sat, Jan 13, 2018 at 2:54 AM, Robinson, Paul <paul.robinson at sony.com> wrote: > Hi Arsen, we are beyond what I understand about how metadata operates. > Maybe Adrian or David knows. > > --paulr > >

Dear All, I bought one IP PHONE from Integrated Networks which was showed to wiki too: http://www.voip-info.org/tiki-index.php?page=Intergrated-Networks I have problems with the Asterisk authentication. It does't want to LOG IN to Asterisk; it always says "LOG ON FAILED". I'm using the IAX2 protocol and all paramters seems to be correct. Does somebody use the same IP PHONE with

Hi Arsen, we are beyond what I understand about how metadata operates. Maybe Adrian or David knows. --paulr From: Arsen Hakobyan [mailto:hakobyan.ars at gmail.com] Sent: Friday, January 12, 2018 12:16 PM To: Robinson, Paul Cc: llvm-dev at lists.llvm.org; David Blaikie Subject: Re: [llvm-dev] StripDeadDebugInfo for static inline functions. Just one update: the function causing the segmentation

Hi Folks, I am running a SPA-3000 behind a legacy PABX on an analog line. I have been able to set up a dial plan that sends outgoing calls out to the appropriate VSP depending on prefix, and that part and the incoming call handling works fine. I am now trying to implement call pickup (dial 6*) or manual call forwarding (flash, dial extension). On the first of these I have worked out how to get

Just one update: the function causing the segmentation fault is the following: 359 void DwarfDebug::constructAbstractSubprogramScopeDIE(LexicalScope *Scope) { 360 assert(Scope && Scope->getScopeNode()); 361 assert(Scope->isAbstractScope()); 362 assert(!Scope->getInlinedAt()); 363 364 const MDNode *SP = Scope->getScopeNode(); 365 366