[Asterisk-Users] spandsp debug or logging

2006-02-23 Thread Anton Krall
Maybe this is a stupid question but how to you enable debubg or logging on spandsp? I see you can do that for RXFAX but when people tell you to enable debug on spandsp, do they mean this with rxfax or how do you do it with spandsp? I have read that sometimes your faxes come as garbage due to

[Asterisk-Users] Which Quad Port FXO is Best?

2006-02-23 Thread John Kelly
I'm looking to handle 3 PSTN lines with my Asterisk server. I have a Digium TDM22B and Sipura 3000. The Sipura works great, but the TDM22B seems to have terrible problems with my board---virtually all peripherals need to be disabled in BIOS, and then there is terrible noise, terrible silence

Re: [Asterisk-Users] What business IP phone to use

2006-02-23 Thread stoffell
On 2/22/06, Clint Sharp [EMAIL PROTECTED] wrote: I had to drop 1.0.1.12 because it has a serious handset volume issue that seems to cut the handset volume in half. Fix one bug, cause another. True, but the latest (beta, okay, but does that matter?) firmware fixes bot and some other. Please

RE : [Asterisk-Users] IAX2 through Shorewall rpoblem

2006-02-23 Thread f6hqz-m
Hello the list, Be carefull to have this rule available at begining of your rules list, because shorewall use the first one matching and stop to check the following. If you have another with a range including this UDP 4569 DNAT before your new one (as UDP 1024 to 65535 for example), it could

Re: [Asterisk-Users] Codec order sent wrong from Asterisk

2006-02-23 Thread Tele Cost Price Reducer
hi Palma, as the SJ initiate the call, it will allways go with GSM Codec as the codec should be identical used on both sides. as you do not have G729 on the SJ, it will never use G729. furthermore, i think that if the GSM will not work, then the second option choosed would be PCMA i hope i gave

Re: [Asterisk-Users] Asterisk hints

2006-02-23 Thread Garth van Sittert
Hi Mike I have build 18 on the IP10's and I have tried call-limit=1 but it still doesn't work. Do the extension phones need to have any settings changed to enable this feature? Here is my sip.conf: [11] callerid=Reception 11 username=11 secret=pbx type=friend host=dynamic

RE: [Asterisk-Users] mpg123 alternative?

2006-02-23 Thread Lee Archer
Check out the musiconhold.conf.sample in the asterisksource/configs folder. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Faris Raouf Sent: 23 February 2006 18:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

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