similar to: switch in extensions.conf

Hi, I have two asterisk machines, chomper and otao. otao is otao.ieworks.net, has a public IP address (66.101.11.61), but no PSTN connections. chomper is at my house, behind NAT, but has a single X100P PSTN connection. I would like to establish two way calling between otao and chomper. Right now, I can call my extension on otao (2101) from my x-lite softphone on chomper, but I cannot call

Hi, I have two asterisk machines, chomper and otao. otao is otao.ieworks.net, has a public IP address (66.101.11.61), but no PSTN connections. chomper is at my house, connected to cable modem behind NAT, but has a single X100P PSTN connection. I would like to establish two way calling between otao and chomper. Right now, I can call my extension on otao (2101) from my x-lite softphone on

<http://voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf%20sorting> This page on voip-info.org describes how it is possible to affect the sort order of patterns in extensions.conf. What is doesn't explain is how asterisk really does sort patterns. How does this happen? Adi

Does anyone know if it is possible to use the following disconnect tone setting with an x100p card? Disconnect Tone: 350@-19,440@-19;4(.25/.25/1+2) This tone was written for a Sipura SPA-3000 for a Panasonic KX-TD1232. The Panasonic does not support disconnect supervision, so this tone is the only thing that will detect a disconnect. It is not a standard fast busy or offhook tone.

Hi everyone, I decided to have a look at SIP & NAT again and I've been at it for a [quite a] few hours but typically nothing is working for me. Actually I'm not sure if SIP and NAT can ever work but some emails on this list do suggest that someone has got it working, once, maybe. I'm experimenting with a ZyXEL 2000W [WiFi Sip phone] which supports "Outbound Proxy",

Hi Ray, I was wondering if the "qualify" option is used [in sip.conf] to keep a connection (from the SIP phone inside the firewall to the Asterisk server outside the firewall) open then would the firewall not allow two way communication without incoming port mapping/NAT (providing that the SIP phone started "talking" first)? I'm not sure about that - I'm being

Hi all I'm trying to understand what the switch => statement in the extension plan exactly does. Unfortunately it seams to be very badly documented. There's allmos noting about it in o'Reilly's Asterisk Book. Nothing in the shipped /doc, nothing on voip-info.org. Even with google I found ony very few infos. So can somebody explain? I got two Asterisk Servers running

Hi all, i need to know how to use the switch statement in extensions.conf to throw calls on a secondary asterisk server. I have been trying all day to be able to make it work but its not working. It displays the following error: pbx_find_extension: No such switch 'master:secret at 192.168.0.7' I have read the help material on voip-info for connecting 2 asterisk servers and followed the

Hello, Sometimes, when a call comes in from the PSTN through our VoIP gateway, the information that is sent to our web page that logs calls includes the original CID name instead of the one that is we expect to be rewritten on the fly using Asterisk's LookupCIDName: ================= ;extensions.conf [internal] exten => group,1,LookupCIDName exten =>

Hello. The procedure so that it works you can find in: http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris k a the files .wav chmod 755 file.wav sox file.wav -r 8000 file.gsm resample -ql chmod 755 file.gsm in extensions.conf xxxx=> xxx,x,playback(file) Ing Javier Rios Ing de Proyectos 04167285748 212 2637246 /2637187 -----Original Message----- From:

I have my server configured to send to send all PSTN traffic to my Cisco AS5300 gateway via SIP. I use the following line in the extensions.conf file to accomplish this: exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@10.1.1.1,240,T) Unfortunately, when I removed the T from the end of the statement, the calls still complete, but they drop as soon as the called party answers the phone. I thought

I have an error sending to a specific fax number. It may be more than one but this is the one I investigated. It seems the delay for the SIP negotiation in T.38 was initiated after 6 seconds, however, our system sent the BYE after only 4 seconds, possibly cutting the call before all the communication necessary for the negotiation was completed. Here is the trace from our provider showing their

Welcome to a new issue of Asterisk Summer News! The holiday season is coming to an end here in Sweden, people are getting back to work and the kids will start going to school next week. Life is slowly adopting to normal and I have to start dressing more towards a businessman than a beach bum. Guess I have to start going to the gym again as well. Anyway, back to the topic. Asterisk and VoIP.

On the extensions.conf explanation page is a mention of the "switch" statement and it refers one to the "connecting two * servers" page. The only mention of the switch statement there is brief and in the example. However, the example seems to have some errors in it. It shows a sample of what's in the extensions.conf file, but it clearly has sections which would be in the

Adam, I thought Andrew Kohlsmith gave the individual good advice without intentionally malaciously spitting in the guys face. For the question, " 'Whats the ' Minimum CPU required for 60 calls? " I think a Pentium 3, high end, which is cheap right now, should do fine, but you will need either 3 T-1s or arrange for the calls to come in via SIP, but you will still need more

Mitel ICP 3300 & Asterisk, Is possible that integration? (C?sar) -----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: Jueves, 25 de Mayo de 2006 03:00 p.m. Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 22, Issue 147 Send

I have a NY 212 packet8 service if you would like to work with me to set this up on my A@H service, I'm happy to test this with you. Cheers, Dean > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Adam Megacz > Sent: Sunday, 7 August 2005 5:29 PM > To: asterisk-users@lists.digium.com

When I switched to 1.8 from 1.4 I am getting this error pbx.c:4055 pbx_extension_helper: No application 'Wait,1' for extension (default, s, 1) http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands This page says its in 1.0 and I dont think has been removed. Did I do something wrong? Everything else seems to be ok. Thanks, Jerry

We are using Bellsouth 8867 phones on our TDM400B FXS lines (asterisk-1.0.3). It has a "Voicemail" light, which appears to be MWI (according to the manual it works with voicemail from the telco that sends a FSK signal). The dialtone stutters when a line has voicemail, so I know that I have the mailbox setting right in zapata.conf, but the light doesn't go on. I am also getting

Hello I have a problem with my MGCP voice gateway. I use D-Link DG104S Boot PROM Version 3.0B38-D Firmware Version 3.0T86-D I tried asterisk v 0.7.2 and I am using latest CVS version now. When I dial a number very fast, or when I use a redial function, my asterisk receives coupled digits. My co-worker called number 245005111, these are a few lines of my debug. The identifier of first digit