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We have asterisk connected to the PSTN VIA TE410P (ANSI SS7 to T1 PRI) cards. I am having trouble completing faxes. Carrier send calls to me using SIP. Any recommendation to have some success with Fax. We trying using T.38 pass through and using G711U codec. Asterisk Version 1.6.1.1 Thanks, Dave -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ATM DS3 from ASI, but we also need them for Augusta and Andover to order DS3s back to Broadway for the collocation equipment moves. Regards, Shaun -----Original Message----- From: David Henning [mailto:dhenning@sctelcom.com] Sent: Monday, June 06, 2005 9:47 PM To: stierney@prairiestream.com Cc: dgeorge@prairiestream.com; dschaub@prairiestream.com; Eric Ryker; Zack Odell Subject: Re: LOA for CFA . . work up "pencil copy" Shaun, Go ahead and complete for Steve Davis (SCTelcom General Manager) to sign. If you want me & Zack to review a "pencil copy", that's fine. When...

Hello! Ater several successful SRTP-enabled calls with SRTP set to Mandatory, asterisk starts to give the following warnings in Log: WARNING[13714] res_srtp.c: SRTP unprotect: authentication failure (continiously) and client hears no sound. After i restart the client program it works fine again for a while. Then the same warning again. Asterisk 1.8.1.1, RealTime engine, sip peer has

My server is being attached all day and fail2ban is not stopping the attack. I updated stamstamp to match fail2ban requirements. [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 handle_request_register: Registration from '"7002" <sip:7002 at x.x.x.x>' failed for '38.108.40.94' - No matching peer found [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830

All- Recently an Asterisk server we host was hacked and used to route some unauthorized calls. We have since improved our security measures, including installation of fail2ban. The interesting thing is the way in which this was discovered. The unauthorized calls were occurring intermittently last Thurs evening thru Sat morning. On Sat morning, some of our employees were attempting to log-in

...ad balance with 2 wan connections > From: Alejandro Imass <ait at p2ee.org> > Date: Sat, December 25, 2010 1:58 pm > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > > On Sat, Dec 25, 2010 at 1:18 PM, dave george <dgeorge at teletoneinc.com> wrote: > > Need some advise or paid help on running asterisk on two WAN connection. ?I > > need load balancing and failover support. > > > > WAN: 1 DSL + 1 Cable ISP. > > > > There are _many_ issues. First outgoing and incoming traffic is...

I have a box (Genband) expecting the following: 100 trying 180 ringing with SDP Or 100 trying 183 with SDP And asterisk is sending: 100 trying 180 ringing 183 with SDP Any way to modify asterisk to send what he is expecting? Thanks, Dave -------------- next part -------------- An HTML attachment was scrubbed... URL:

TE410P card down. I have three (3) TE410P in one machine running asterisk with SS7. My problems started last week when one of my cards started switching to E1 every time after reboot. I set the following in dahdi.conf and that solve the problem. /etc/modprobe.d/ options wct4xxp t1e1override=0x00 Now all 4 ports on that card is down with Red Alarm. I tried rebooting the machine and

My users dial *120 get to an IVR menu that plays their balance and then ask them for a voucher. Ater the balance is played and the request for the voucher is played the user don't hear any other audio from the asterisk box. I can see the asterisk server playing the files to ask for the voucher again but the user cannot hear any thing. Has any one seens this issue with IVRs. I notice a