Re: [asterisk-users] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for public_ip:2049

On 10/13/2010 12:09 AM, Paul Belanger wrote: > On Tue, Oct 12, 2010 at 8:08 AM, Jonas Kellens > wrote: > >> [Oct 12 14:03:32] DEBUG[9064] chan_sip.c: Trying to put 'SIP/2.0 401' onto >> UDP socket destined for public_ip:2049 >> >> Is something failing, or is this just informative ? >> >>

Re: [asterisk-users] sound file debug

On Tue, Oct 12, 2010 at 4:23 PM, Danny Nicholas wrote: > dollars.gsm: data > dollars.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, > mono 8000 Hz > dollars.WAV: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, > mono 8000 Hz > > Can't be 100% certain on #2, but it

Re: [asterisk-users] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for public_ip:2049

On Tue, Oct 12, 2010 at 8:08 AM, Jonas Kellens wrote: > [Oct 12 14:03:32] DEBUG[9064] chan_sip.c: Trying to put 'SIP/2.0 401' onto > UDP socket destined for public_ip:2049 > > Is something failing, or is this just informative ? > No, this is a debug message. Unless you are trying to solve a probl

Re: [asterisk-users] Receive Call from unknown user

On Tue, Oct 12, 2010 at 10:05 AM, Stefan Schmidt wrote: > so if you dont know someone in china, it would be a good idea to block > this AND set allowguest=no to prevent this in future. > And firewall your Asterisk box. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybea

Re: [asterisk-users] sound file debug

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton West Sent: Tuesday, October 12, 2010 3:15 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] sound file debug You have two separate

Re: [asterisk-users] sound file debug

You have two separate problems here: (1) >dollars.WAV: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 8 bit, >mono 8000 Hz You should have generated this with 16-bit resolution, like all the others. (2) Not sure about the cents - sure it's coming out as 16-bit? Is the file in the right

[asterisk-users] sound file debug

Hi gang, I have a "fun" one for you. I'm not getting the quality of sound I want out of GSM, so I'm trying to make my files into .WAV (.wav) format. Here is the "file" output for 5 files: file *.WAV cents.WAV:RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 80

Re: [asterisk-users] user number in conference

On Mon, Oct 11, 2010 at 6:14 PM, Daniel Knoll wrote: > Hey, > i forgot to ask, how can i get the user number from a caller he is in a > conference, i don't find a variable to us this for the current channel. > Only the command "meetme list " shows the usernumber, but i can't use > this output.

Re: [asterisk-users] how to fake asterisk register?

Check the Asterisk option autocreatepeer. See http://gnuradio.org/redmine/wiki/gnuradio/OpenBTSThe_use_of_autocreatepeer=yes for an example. This is specially made if registration is done by a real sip register like Kamilio. \Erik On 12 okt 2010, at 16:10, Borin wrote: > Hello, > > I am wo

[asterisk-users] REFER method

Hello, Is there a possibility for asterisk to work out REFER messages in the dialplan? like INVITES. I need it to distinguish forwarded calls from all the other calls. -- _ -- Bandwidth and Colocation Provided by http://www.api-d

[asterisk-users] how to fake asterisk register?

Hello, I am wondering how I can make asterisk think that the user is registered on it..Scheme is the following: user register>kamailio (put data in location table)--asterisk All users are registered on kamailio and I want to duplicate that info on asterisk. If a call comes to kamailio

Re: [asterisk-users] Receive Call from unknown user

Am 12.10.10 07:57, schrieb Malvin Rito: > Hello List, > > I have noticed for the past few weeks that someone from an unknown IP is > trying to make a call to my Asterisk box, below is the sample content of the > log file. Sometimes the calls are being made every seconds. > > Is my system being ha

[asterisk-users] src_mysql problem

Hello, I am using 1.6.2.9-1+b1 asterisk.with cdr_mysql. Everything seems workging correctly except cdr logs. It fills up all data when a call established except src and clid Wht can cause this and where should i check?? -- _ --

Re: [asterisk-users] Create channel bank with TDMoE

Tank, I want create channel bank with ip04 (Blackfin+uClinux+4FXO/FXS+1NIC). I think than zaptel can do it with TDMoE and DACS+RBS ,... Please help me. regards. On 10/11/10, Gareth Blades wrote: > Karim Davoodi wrote: >> Hello, >> I want to create channel bank in this case: >> >>"channel ba

[asterisk-users] Application Map Not Working

Hi, I'm using the applicationmap in features.conf to allow the user to press *1 and run a macro which records using MixMonitor. All was fine in our office (behind a nat). But when I took the sip phones to an end user (also behind a nat), I found that *1 didnt work for outgoing calls. But for s

[asterisk-users] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for public_ip:2049

Hello, what does this message mean ? [Oct 12 14:03:32] DEBUG[9064] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for public_ip:2049 I find this in my debug log file when "core set debug 25". Is something failing, or is this just informative ? Kind regards, Jonas. -- _

[asterisk-users] rtpip patch

Hello *, is the rtpip patch still valid for asterisk 1.6 (with some code changes, obviously)? https://issues.asterisk.org/view.php?id=8161 Or, in asterisk 1.6 there is an alternative to using it? This is the difffile I produced for chan_sip.c in asterisk 1.6.2.11 --- chan_sip.c 2010-10-12 13: