search for: djimenez

...t, 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 administrative assistants who monitor calls for 3 or 4 other people. It'd be nice to have two line instances for them, and one for the person(s) whom they assist. Contact me: djimenez at pobox.com if you're interested in making this happen. -- Daniel Jimenez <djimenez[at]pobox[dot]com>

...etup 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>

...eard 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>

...e 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>

...s 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>

Could anyone who has successfully configured Asterisk to use g729 to conference 10-20 people please post their configs. I purchased and successfully installed 2 g729 licenses and but when I dial into my conference number on the Asterisk box from a SPA-2000 set to allow all codecs, it always appears to connect using ULAW. My iax.conf file includes the following under the general section

I'm trying to see if this is even possible. When you dial ext 2000 I want it to ring my sip phone for 20 sec then call my cell and let it ring for 10 sec if I do not pick up the call on my cell I would like it to go back to * and leave a voice message for me. Here is what I have so far in my extensions.conf Everything works except the call will not go back to * after the 10 sec rule has

...n of code for dose response analysis that would be compatible with a Linnux box? I did not see a specific add in that was quantal dose response - Design looks close,,, any help out there? Desmond R Jim?nez Ph.D Scientist AgraQuest Inc. 1540 Drew Ave., Davis, CA 95618 (530) 750-0150 ext. 125 djimenez at agraquest.com www.agraquest.com Innovative natural product solutions for pest management This email message and any accompanying document(s) contain confidential information that is only for the use of the intended addressee. Any unauthorized review, use, disclosure, or distribution is st...

...t of time later. Sometimes only one will go away, sometimes one will stay. I don't know if this is just a delay in registration or if there is a problem. All of my phones are behind NAT. The Asterisk server is not behind NAT. I am running CVS head from the 19th. TIA, -- Daniel Jimenez <djimenez[at]pobox[dot]com>

...(0,3) exten => 0,104,Goto(0,3) This should call 713-555-1212. If there are no ZAP lines available it should kick back around and play music on hold until a zap line is available, correct? I'd like the music-on-hold to be continuous until the Zap line answers. TIA, -- Daniel Jimenez <djimenez[at]pobox[dot]com>

I've configured voicemail.conf to delete voice mails after they are emailed. The email work ok but the message don't delete. The config is as follows: [default] 3000 >= 1111,CS,cs@x.com,,delete=yes Thanks, Chris

...hole on the firewall (5034 is open, UDP). Everything works great except MWI never comes on. The firewall is a PIX firewall. The only ports open to my asterisk box are 22(ssh) 80(www) and UDP 5034(SIP). It appears SIP works great because the calls function fine. Thanks!, -- Daniel Jimenez <djimenez[at]pobox[dot]com>

Hey all, Did I see something on here about using an AGI script to do reverse lookups via anywho.com? I have a PRI that only gets caller-id number and no Alpha. TIA, -- Daniel Jimenez <djimenez[at]pobox[dot]com>

Hello, I've recently purchased Adit 600 with 3FXS and 1FXO to be connected to my * server via T100P card. This is the output of "status equipment" command in the Adit600: For some reason the FXO card is seen as FXS, why? Is it ok? On the card it is written "FXO". Regards, Shlomi Bachar -------------- next part -------------- An HTML attachment was

Hello, I've been playing around with two generic X100P analog cards to create a proof-of-concept system before we go ahead and hook up a PRI. I'm running into a reproducible problem with sound quality of bridged calls, and am hoping someone will be able to point me in the right direction. I have in my dial plan a _9. extension so outgoing calls can be made... the first thing is

Hi, i want to use soxmix to record some calls in my PBX. If i use soxmix on my linux shell it works so i can mixed two calls into one consolidated call. I want to do the process automatically since extensions.conf but it doesnt work. My extensions.conf looks like this: exten => 407,1,SetVar(MONITORDIR=/var/spool/asterisk/monitor) exten => 407,2,Monitor(wav,${TIMESTAMP}.${CALLERIDNUM}.wav)

Hi, We are having an issue here. It seems that whenever we initialize Asterisk on our network, the router that the Asterisk server is connected to (Cisco 7200) crashes and loses it configuration. This has happended five times and each time we have tested it, it is always when Asterisk starts up. Has anyone else seen this problem? It is very odd because this is a very good router and we

Over the last couple weeks I've tried everything I could get my hands on in an attempt to get rid of my echo problems. Using a CVS checkout of just yesterday, I've tried every echo cancellation routine in zconfig.h (including Mark2 w/Aggressive) , as well as the echotraining=800 mentioned on this list just last week. While some things worked better then others, I would consider none