similar to: call parking timeout

Hi * users, I want to know if STUN suport is available with Asterisk. Kindly let me know. I have posted this also in DEV list but none replied to me. thanks, Somesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050808/e26855c9/attachment.htm

Tie line is type em_w (old school stuff) to TE110P. Phone rings on the asterisk side, but the calling party does not hear the ring through sound. If I pick it up within the first two rings it goes through and I can talk otherwise our old switch drops the call. Anyhow...here is my config if anyone can shed some light on it. It used to work with HEAD a few weeks ago. -Matt

Hello, I am trying to let someone know that is being called from a specified location. For that, the command: exten => _107.,1,Dial(SIP/SIPGW/${EXTEN:3},30,A(Anounce.gsm)) should let the called person hear Anounce.gsm as soon as he/she answers. (Only calls with prefix 107 are given this notice). The call proceeds fine, but no one hears AnounceSPF.gsm. I tried putting this file in every

I just bough a Sipura SPA-2100 to use with Asterisk. When I use the analog handset plugged into the SPA-2100, the person on the other end can hardly hear me. I check the SPA-2100 setup and their is no mic/spk gain control. Is this a problem with the SPA-2100 or with Asterisk? Any way for asterisk to compensate for the poor audio level (if the problem is the SPA-2100)? Thanks, Mike

The cheap ones on EBay won't work with the SC420 server. I have one and can't make any of the clones work. I do have one TDM40B card for analog stations that works well. The problem with the SC420 is that it won't let you set the interrupts yourself and you end up with interrupts being shared. =============================================================== Message: 26 Date:

I've just gotten off a skype conference call and it pisses me off that the quality of skype is higher than my asterisk calls. Is there such a thing as a super high bandwidth codec? In a situation that you have the bandwidth to share is there something that I can use for important calls when the situation warrants it? TIA, Dean -------------- next part -------------- An

is it possible to register asterisk in a sip proxy as if it were a terminal (like a cisco ATA)? how? Thanx Jenna ;) ___________________________________________________________ 1GB gratis, Antivirus y Antispam Correo Yahoo!, el mejor correo web del mundo http://correo.yahoo.com.ar

Hello, I sent this request to my local linux user group: <http://lists.luv.asn.au/wws/arc/luv-talk/2008-02/msg00000.html> Unfortunately it looks like nobody is able to answer, so I will try here: Does anybody know of any sane solution for (globally) managing the configuration of printers on a large number of Windows computers? My current solution is 1. create samba server with printer

Hello, I have a TDM400 with 2 FXS modules and 2 FXO modules, as follows: Zap/1 - internal phone Zap/2 - internal modem Zap/3 - exchange Zap/4 - exchange Recently I upgraded zaptel (and kernel modules) from 1.0.9-1.1 to 1.0.9.1 and upgraded asterisk to 1.0.9.dfsg.1-3, which are Debian versions for Debian/sarge of the packages I believe the Debian maintainers created (not in Debian) The above

Dear R helpers,   I understand that this is absolutely unprofessional on my part and this group doesn't entertain such things. I have been associted with this group since last 1 and half years and have been immensely benefited by the noble service rendred by many R helpers.   So I take this opportunity to thank all of you and wish you all   "MERRY CHRISTMAS".   I sincerely apologize

Hi, Everybody, I can modprobe TDM400P card without feeding power supply , so what is the purpose of providing power supply in that card. Can any body tell me Regards Lokesh Portugal mail -lokeshkumar80@yahoo.co.in ____________________________________________________ Send a rakhi to your brother, buy gifts and win attractive prizes. Log on to http://in.promos.yahoo.com/rakhi/index.html

Hi peoples Can anyone tell me if the Zultys Zip 4x5 supports iax protocols or if they have configured one before for iax. If you have a sample config file that would be great. Any assistance would be nice Scott -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050811/6c243053/attachment.htm

Hi; Do you know where the addmailbox script goes? Now in CVS, it shows revision 1.3 date: 2005/08/02 11:57:06; author: markster; state: dead; lines: +0 -0 This script is now useless... ---------------------------- So any new tool to create voicemail folder? Thanks Kun

I am using asterisk (2 week old CVS) am for the first time have been starting to experiment with busy and congestion. At this point I am only using sip endpoints PAP2-NA devices. All testing of this is being done on a local network. my test extension looks like this: exten => 7777,1,Answer exten => 7777,2,busy(35) exten => 7777,3,Hangup Or like this: exten => 7777,1,Answer

We have recently started routing about 3 PRI's worth of traffic thru our asterisk box. The text on the console now flys by so damn fast, I can't really see what the heck is going on. Even with verbosity 0 and debug 0 it is still so fast. Is there some way I can attach to the console in a way that will allow me to grep or otherwise filter the text so I can focus on something in

Anyone having trouble parking calls? I haven't tried it in a while, but it seems to have stopped working. If I dial 700, I get a invalid extension. I have "include => parkedcalls" in the correct context, and I can dial 701, which tells me no call is parked there. Any ideas? Parking.conf is stock.

I am trying to configure the FXO port on a Sipura-3000 for use with Asterisk. When I connect to the Sipura to dial out on the PSTN line connected to the Sipura's FXO port, it gives me the dialtone of the PSTN line and then I can hear the DTMF for the number I dialled beforehand. It does work but the customer perceives this delayed second DTMF feedback as "unprofessional" and the

A fairly serious question: how much can we (as users) donate to get the Google ads off the Wiki? I guess this does not give so much revenue, and it is really distracting. So, if we can match the annual income of Google Ads on the Wiki, I think I (and others) are willing to compensate this with donations. I have seen some potential Dutch users getting turned away seeing the main CentOS site

We have been running into problems here, we have 2 PRI's when they fillup, All channels in use, and we dial more calls asterisk becomes unstable and crashes alot. We are currently on: Asterisk CVS-v1-0-10/26/04-09:35:31 built by root@localhost on a i686 running Linux I know I need to upgrade. Is this a know issue?? Kyle

I'm trying to get this zap dial to work. I want to send DNIS and ANI to other system (ZAP/g2) at answer, while the caller hears ring (RBT). I hear the RBT, but callerid and exten is not sent to other T1 - The ZAP/g2 T1 is standard D4, SF, E&M Wink Start. - At ZAP/g2 wink, asterisk should send DTMF "*ANI*DNIS*" exten => _XXXX,1,NoOp,${CALLERID} exten =>