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Hi, I asked yesterday about a problem with x306 and IRQ sharing, didnt get much info, now, i was playing with lspci, and see something strange, lspci -v shows me the TDM400P card is on IRQ 7, and the SCSI card is also on IRQ 7, lspci -bv (from the man - b - shows "bus-centric view, as seen by the BUS and not by the kernel) shows me the TDM400P is on IRQ 5, why does the kernel puts it on

I have been impressed with the quality and meagre bandwidth of the G729 codec from Digium. I am in a testing phase of our roll out, we are using 5 Asterisk PBXs in various countries to provide connectivity for our employees, owners and family. As we are testing, and our setup is somewhat complex due to the peculiarities of our connectivity, there has had to be a lot of changes to servers, cards to

Hi all, Ok, I've been bashing my head for a few hours now on this, trying to figure out if I've done something wrong, but everything seems to me hunky-dory. So here's the deal: 1. I've compiled the spandsp 0.0.2pre10 source code successfully and also the asterisk application associated with it. 2. Receiving a fax at asterisk works fine, at least appears to be working

Hi All, I've been fiddling around with the RealTime configuration. For SIP and IAX it's really cool, and the switch thing is cool too. But I've tried performing a GOTO from one RealTime context, to a second RealTime context. That didn't really work. Any idea how to make it work ? apart from simply defining contexts in /etc/asterisk/extensions.conf manually mapping the

Hi All, I've been observing a very odd behaviour of Asterisk, when relating to SIP connections. Here's the scenario: Ast1 is an Asterisk box originating calls via a predictive dialer Ast2 is an Asterisk box connected to 3E1 circuits Ast1 originates calls to Ast2 via SIP, in order to utilize the PSTN lines. (There is a reason I'm using SIP here, so please don't say:

Hi all, I'm experiencing weird cutoffs on TE110P. All cut offs are pre-seen with an indication 5 coming from the PRI. I've talked to the telco, and they indicated that they don't see any issues. I've also modified the sync source to be the telco, and that didn't solve the problem either. Any ideas anybody ? Nir S

Hi All, I'm trying to connect to a SIP carrier who never connected with Asterisk. I managed to connect with a sipura phone or a grandstream, no problem. When I configure asterisk, I'm able to send out calls to the carrier no problems, however, receiving calls doesn't work, and I keep getting the following messages: <-- SIP read from 69.xx.xx.xx:5060: INVITE