similar to: FWDout credits sharing

It looks like DTMF codes are not properly transmitted by bellster. For example, you can try the toll-free number 33800123456, which asks you to press *. When I tried that yesterday, the connection got dropped. Sam -- Samuel Tardieu -- sam@rfc1149.net -- http://www.rfc1149.net/sam

I have a problem with ZAP interface bridging in France (FXO interface): hangup is detected through a busy tone (no polarity inversion or whatever). When I dial out from a zap line when I receive an incoming call on another zap line (for example to redirect calls to my office when I'm not home), caller hangup is not detected because Asterisk seems to put itself out of the voice path because it

I am trying to register my Asterisk server to a SIP server which doesn't accept an Expires: field smaller than 1800 seconds and indicates it correctly with a Min-Expires: in an error response when Asterisk tries to use its default of 120 seconds. Is Asterisk supposed to honor this field and retry with the proposed minimum Expires: field? It looks like it doesn't, and I had to change the

Hi. Is there a 802.1x implementation (client and server) for FreeBSD -STABLE? Sam -- Samuel Tardieu -- sam@rfc1149.net -- http://www.rfc1149.net/sam

Many friends of mine asked me to describe my home Asterisk setup. I've done that at: http://www.rfc1149.net/blog/index.php/mrhyde/2005/03/23/asterisk_build_your_own_pbx (or use the shorter http://tinyurl.com/5s79m URL) Sam -- Samuel Tardieu -- sam@rfc1149.net -- http://www.rfc1149.net/sam

Hi. I am looking for an IPv6 capable CVSUP mirror. I found a discussion from one year ago where it was stated that CVSUP was not IPv6-capable. Does anyone know if this has changed? Sam -- Samuel Tardieu -- sam@rfc1149.net -- http://www.rfc1149.net/sam

Steven P. Donegan wrote: > I don't want to be negative here, but I do believe people who go to do this know the potential risks they face. In many countries (4 of which I have direct, although several year old experience with - all in Asia) taking a local phone line and attaching asterisk to it and gatewaying traffic from other countries will be considered to be 'theft' by the

Hi. I have just added two FXO cards in my PC: - Zap/1 is my France Telecom telephone line - Zap/2 is my Free telephone line (Free is an ADSL provider which provides an additional line using VOIP, but this line is only accessible as an FXS, no way to use it directly in H.323 or SIP) In incoming mode, those two lines ring different extensions in my phone installation. In outgoing mode,

Sam> In France, the second most important ADSL provider (named "Free") Sam> offers a phone line (which uses VoIP but can only be used as a FXS) Sam> with unlimited free calls to landlines. I also have Free ADSL in Paris, and would very much like to get their VoIP working natively with Asterisk. Free assigns each user both a public (for Internet access) and a private (for VoIP

OK, I have done all the stuff at my end and at Bellsters end to add 21 new area codes (all of california) to the Bellster dial plan. Pretty cool deal! I hope others go for this quickly - as it could be a really nice co-op. I do suggest to Jeff - do some sort of calling trunk -vs- routed trunk match to make sure that someone can't run their credits sky-high by making calls through

I cant seem to be able to figure this out. As much as I can tell it is a codec problem. I can dial out to 612@fwd.pulver.com and the "Call Me" test there rings my phone. However when the callee endpoint answers, there is a failure to translate: Outgoing Call for 612 612 is not a local user -- Called 612@fwdpulvercom No path to translate from SIP/fwdpulvercom-dd5a(2) to

As far as I know it's not legal to join bellster in Israel. It means that you're reselling the minutes you buy from the telco company. It also means that you need a permit from the Israeli ministry of communications cause you're acting as an international call provider. Can't be done here. -----Original Message----- From: Geoffrey S. Mendelson [mailto:gsm@mendelson.com] Sent:

>Shoval Tomer wrote: >> As far as I know it's not legal to join bellster in Israel. >> >> It means that you're reselling the minutes you buy from the telco >> company. > >Wouldn't you need to be selling them to be reselling? > >Does that make DISA illegal, and VoIP connections between offices if you >dial out the other end? Well, a thing

Hi, I have a problem compiling openssh pre 1.12 on solaris 2.5.1 platform with gnu gcc 2.95.2 u_int32_t is missing somehow and i cannot find any includes which define it. gcc -g -O2 -Wall -I/usr/local/ssl/include -DETCDIR=\"/usr/local/etc\" -DSSH_PROGRAM=\"/usr/local/bin/ssh\" -DASKPASS_PROGRAM=\"/usr/local/lib/ssh/ssh-askpass\" -DHAVE_CONFIG_H -c authfile.c

Hi, Here's what I get (pre15 with __P(x) x fix) : ar rv libssh.a ranlib libssh.a gcc -o ssh -ldl -lsocket -lnsl -lz -lcrypto -L/usr/local/ssl/lib -lssl -lcrypto Undefined first referenced symbol in file main /usr/local/lib/gcc-lib/sparc-sun-solaris2.5.1/2.95.2/crt1.o ld: fatal: Symbol referencing errors. No output written to ssh

FYI It looks like FWD is looking for value added service ideas for free as a volunteer. I think it will fail but we shall see. I really don't get the nerve of them (Free World Dialup has changed it's name to FWD) to ask for free ideas and development on a non-free service. Maybe if they can come up with a killer app and people will adopt it, then it might work, but then again, people

I didn't get any response at all to my last "request for status" on IAXTEL. So, when this happens, I attribute it to one of a number of things: 1. No-one knows. 2. No-one cares. 3. Everyone knows, but are too busy to reply. At any rate, my investigative side kicks in and I began searching thru the digest's I've gotten, looking for references to IAXTEL. Mostly it is

The agenda for Astricon Europe in Madrid June 15-17 is now coming together. It will be an international conference, with speakers from both USA and Europe. Last year, we had over 25 nationalities participating in the first Astricon - the Astricon where Mark released Asterisk 1.0 on the conference floor, during his keynote! Many active members of the Asterisk community talks at the conference, one

merlin# rm GENERIC merlin# cd /usr/share/examples/cvsup/ merlin# cvsup stable-supfile Connected to cvsup.uk.FreeBSD.org Updating collection src-all/cvs Checkout src/sys/i386/conf/GENERIC Finished successfully merlin# cd /usr/src merlin# make buildkernel KERNCONF=GENERIC -------------------------------------------------------------- >>> Kernel build for GENERIC started on Sat Jul 26

I have an asterisk servicer (1.0.5) with 3 X100P cards. Everything is working fine but two days ago I implemented call forwarding using the example from voip-info wiki. Now when I enable call forwarding on my phone and a call comes in it gets redirected to my cell and everything is apparently working. The problem is that when we hang up both Zap interfaces (the one where the original