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http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry From the WIKI: Contributions Manager: Daniel Jimenez (cuban) Bounty: $50 USD Date opened: July 10, 2004 Contributors: cuban ($50) Detail Yes, Yes I know you could do all sorts of fun with the dialplan to produce a similar effect, but I still would like to be able to do this. Plus it's easy money :). I have some users with a 7960 who are admi...

...ering if I setup SIP on the as5300 I can have asterisk use the free channels for dial out. I'd still have to use my TDM04B for incoming calls, but at least I can expand my outgoing. Anyone done anything like this before? I've never messed with VoIP on Cisco equipment before. -- Daniel Jimenez <djimenez[at]pobox[dot]com>

...? I've heard ISDN4LINUX devices suffer bad echo but chan_capi works great. All the chan_capi cards I find though are for overseas (ie europe etc). Would I be better off looking at a fractional PRI? I'm only using 4 lines right now. I think a fractional PRI would be over kill. -- Daniel Jimenez <djimenez[at]pobox[dot]com>

https://bugs.freedesktop.org/show_bug.cgi?id=92961 Bug ID: 92961 Summary: Xorg freezes (only mouse and ssh are still working) Product: xorg Version: unspecified Hardware: x86-64 (AMD64) OS: Linux (All) Status: NEW Severity: critical Priority: high Component: Driver/nouveau

...oes that mean anything? genzaptelcong doesn't generate any config for the Astribank. Can anyone send a copy of their working Astribank settings? I am concerned about the span used by the Astribank. I just cannot find anything that gives me a clue... Thanks in advance. Regards, -- Andres Jimenez GPG : http://www.andresin.com/gpg/gandresin at gmail.com.asc

...rk with a DHCP server on a BSD machine (using bootpd) and at least I got to the welcome screen (then the phone hanged and the reboot keystroke did not work)..but I think that last thing is just an issue with everything else not being setup on the phone itself. Any ideas? Thanks, Eduardo Jimenez Software Engineer Job Performance Systems, Inc Ph: 703-683-5805 x 210 www.jps-usa.com <http://www.jps-usa.com/> Also, I unpacked a phone, played with it for a minute and now that phone does not seem to boot. Tried the 4 6 8 * combination for about 10 secs and did not do anything..is...

.... let say I want user 132 (that uses extension 132 in our system) to be able to lock his phone (located in a publicly accessible office). Could he dial an special extension (i.e. 9999) and Asterisk will drop any call until another special extension (i.e. 8888) is dialed? Suggestions? - -- Andres Jimenez -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.3 (GNU/Linux) Comment: http://firegpg.tuxfamily.org iD8DBQFGxX6A8SZxpGYWwpYRAtojAJ4yKE77nv9rpkoXXr1i4SOiLPb7JACgug+7 64yg8fCDRdnmeZFmmpGynwQ= =eCWP -----END PGP SIGNATURE-----

...just set the dial plan for : (not actual coonfig) exten=> 111111,1,Dial(Zap/41) exten=> 222222,1,Dial(Zap/42) Do I need any other piece of software? I know 1.6-beta is capable of managing faxes properly, but I won't upgrade my * if any other option is available. Thank you. -- Andres Jimenez GPG : http://www.andresin.com/gpg/gandresin at gmail.com.asc

...ocancelwhenbridged = no facilityenable = yes musiconhold = default ;overlapdial = yes overlapdial = no immediate = no txgain = -4.0 rxgain = -4.0 signalling = pri_cpe channel => 1-15 ;channel => 17-32 channel => 17-24 ;toneduration=100 toneduration=300 ;relaxdtmf=yes Thanks, -- Andres Jimenez GPG : http://www.andresin.com/gpg/gandresin at gmail.com.asc

...that feature and love it. Their VM prompt would say: "Hello, My name is blah blah. I am currently unavailable. If you would like to speak to an operator press 0 now, otherwise leave me a message". Extension 0 exists, but dialing it during a VM prompt does nothing. Thanks, -- Daniel Jimenez <djimenez[at]pobox[dot]com>

...nds of times and never stops. chan_sip.c:7467 sipsock_read: Failed to grab lock, trying again... During this period, I can make no SIP calls what-so-ever. The only way I've been able to stop it is to killall -9 asterisk. Doing a restart now doesn't respond. Anyone know why? -- Daniel Jimenez <djimenez[at]pobox[dot]com>

Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo cancelation and a quad FXO card. We have 4 analog lines, one of which is a Cellphone line for least cost

...cluded 3 patches that updates documentation. This completes a 5 patches set. If you prefer me to resend all of then as one patch, attached, discussing or whatever, your're welcome. Best regards, vicente >From 7cec3ad78c8454408c8b6a1950d441e02d56d138 Mon Sep 17 00:00:00 2001 From: Vicente Jimenez Aguilar <googuy at gmail.com> Date: Fri, 23 Jan 2009 00:57:48 +0100 Subject: [PATCH] tftp tsize option don't needed anymore starting at version 3.70 --- doc/pxelinux.txt | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/doc/pxelinux.txt b/doc/pxelinux.txt index 8d...

Hil list! I'm trying to authenticate Active Directory Users via freeradius. I can do it in a general case (user and domain) without problem. Now I have to do it restricting the authentication to the members of a group. I can exect the script (as is put in radiusd.conf) correct from the command line: Deb:~# /usr/bin/ntlm_auth --username=javi2 --require-membership-of='AAMM\MyGroup'

Hi @ all, i set a server to a costumer of mine with a TE207P for use with 2 E1 Lines. I set them together into one group in zaptel/zapata.conf The point is now, the customer has a free-volumina of 60k minutes per month, per line. How can i make a kind of load balancing, that both lines will be trafficed the same way ? I read something about DIAL(Zap/r1/.) for using round robin, and

Dear all, I have small question in sip.conf I added [service] type=friend ;username= ;secret= qualify=900 host=X.X.X.X dtmfmode = rfc2833 disallow=all ;allow=g729 allow=gsm allow=alaw allow=ulaw and I can proccess incoming call from soft phone only I calling on number that is used in extensions.conf(in example below it is 1) exten => 1,1,Answer; exten => 1,2,Playback(hello-world,skip);

On Thu, July 21, 2016 07:56, Jose Maria Terry Jimenez wrote: > El 21/7/16 a las 8:53, geo.inbox.ignore escribi?: > >> greetings to all. >> >> centos = 6.8 current >> system = toshiba l455d-s5976 laptop >> >> a new problem has developed after 1st updating of 6.8. >> >> regular user is not able to op...

...eproduce a plot with three different horizontal axes. I know how to make plot with two different horizontal axes (one above, one below) using axis(): axis(1, ....) axis(3, ....) However, I don't know how to produce two axes on the same side of the plot. Any pointers or examples? -- R. Jimenez

...her method that won't be broken when the asterisk packages are updated. Would anyone consider just install everything from source (branch 1.4) as the best option? I would like to keep an easy upgradeable system like Debian packages, but could use source code if necessary. Cheers, -- Andres Jimenez GPG : http://www.andresin.com/gpg/gandresin at gmail.com.asc

Hello This UPS is compatible ? http://www.voz-ip.com/apcbackupscs500-p-223.html Thanks -- Mikel Jimenez Fernandez Irontec, Internet y Sistemas sobre GNU/LinuX - http://www.irontec.com +34 94.404.81.82