similar to: SipSak: error: this FQDN or IP is not valid: voicegw

Hi! Are you also getting in trouble while trying to register in Broadvoice? Cumprimentos / Best regards, Helder Rog?rio __________________________________________ Microrede - Tecnologias de Informa??o, Ltd. http://www.microrede.pt *** ? There are only two types of people in the world, those who have lost data and those who will. ? -- Richard Nixon

Hi, I've been loosing my mind with NAT and read that IAX doesn't have problems about nat. Does anyone knows about hadware (routers and etc) support IAX? Best regards helder -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041230/f689bdde/attachment.htm

Hello, Is there an utility for asterisk for codec conversion? I tried google but i haven' got anything. I am trying to initiate a call with G711 codec to asterisk and i would like asterisk to call a gateway with an g729 codec, therefore making a codec conversion from g711 to g729. I know chan_oh323 does it by specifying the OUT_CODEC variable, but chan_h323 does not. And i was wondering is

Hello, I'm trying to use SipSak to check if my Asterisk server is available/running with the following : sipsak -vv -s sip:username at sip.domain.tld -c sip:username at sip.domain.tld --password guessthis --hostname XX.XX.XX.63 The SIP OPTION is received by Asterisk as follow : OPTIONS sip:username at sip.domain.tld SIP/2.0 Via: SIP/2.0/UDP

I'm using sipsak to send messages to Snoms in my subnet. At work, works fine: sipsak -M -O desktop -B "foo" -s sip:1001@192.168.1.220 -H 192.168.1.46 displays "foo" on the Snom display On my home LAN (AAH 1.5, Snom 190 3.60s, switched 100, no VLAN, no routing) the same command (modified for my LAN) always yields: (type: 3, code: 3): from 192.168.171.8 at the console

On Fri, 20 Feb 2015 15:07:35 -0500, Andres wrote: > This is showing nothing so I don't think your test message even made it > here. I think it looped in the 'doge' server. I was wondering the same thing :) in tleilax, I looked in /var/log/asterisk/messages and see: [Feb 20 15:13:19] VERBOSE[3661] chan_sip.c: [Feb 20 15:13:19] <--- SIP read from UDP:192.168.1.3:38154

On Fri, 20 Feb 2015 08:46:13 -0500, Andres wrote: > A "sip set debug on" will give you more info on why you are getting the > 404. It probably has to do something with your context/dialplan. on tleilax: tleilax*CLI> tleilax*CLI> sip set debug on SIP Debugging enabled tleilax*CLI> on doge: thufir at doge:~$ thufir at doge:~$ sudo sipsak -vv -s sip:devries at

What's the difference between user "123" and "devries"? Based on the output here, they seem the same..? tleilax*CLI> tleilax*CLI> sip show users Username Secret Accountcode Def.Context ACL Forcerport 201 password 201 default No Yes 123

I run configure in a build folder using CC=gcc CXX=g++ ../configure --prefix=/project/scratch/packages2/clang \ --enable-targets=host --enable-shared The configure runs fine but when I type make I get the error: make[1]: Entering directory `/project/scratch/tmp/llvm-3.3/build/lib/Support' llvm[1]: Compiling APFloat.cpp for Release+Asserts build In file included from

On Mon, Jun 24, 2013 at 6:17 PM, Jorge Rodrigues <skeept at gmail.com> wrote: > Norm, > > thanks for the help. Applying the fix solves the issue I mentioned but now I > have more issues. > > I can install clang, but when running I cannot compile and link files. > If I compile with -c flag it works but compiling the following x.c file > gives an error: >

On Jun 24, 2013, at 4:18 PM, Stefan Teleman <stefan.teleman at gmail.com> wrote: > I'll file the bug about the endian stuff, but we'll also have to have > an <endian.h> in Solaris as well. Do you still have issues after r182419? commit 01ef4f6982451e6a7c00a713d9ae677d3a15d042 Author: Jakob Stoklund Olesen <stoklund at 2pi.dk> Date: Tue May 21 13:36:13 2013

Hi guys, I''m just beginning to use RSpec and I ran into the issue of testing a named_scope. I''m not actually trying to test its behavior, as it''s not my code, but I wanted to test at least that it''s defined. I tried doing this: describe Post, ".most_recent" do it "should be defined" do Post.method_defined?(:most_recent).should be_true

try "sipsak -M -O desktop -B "foo" -s sip:<user>@<registrar> -H <ip of registrar>" the trick is to specify the "-O desktop" parameter + the "-H <ip of registrar>" parameter. Sipsak fakes the host-header of the registrar so that the Snom thinks it is coming from your Asterisk server, then lets the message through to the

Norm, thanks for the help. Applying the fix solves the issue I mentioned but now I have more issues. I can install clang, but when running I cannot compile and link files. If I compile with -c flag it works but compiling the following x.c file gives an error: x.c: int main(void) { return 0; } > ./clang x.c /project/helder/scratch/packages2/bin/ld: unrecognized option '-C'

Hi, I''ve a working xen 3.0.2 booting a gentoo linux on Dom0. I have setup a Windows XP in a DomU OK with no problems except the network!! My problem is that I wanted to give a working network interface on the DomU and the best I could. making the windows detect an network adapter, was putting: vif = [''type=ioemu''] I also tryied: vif = [''type=ioemu,

I would be very interested in using sipsak for something like this. What have you tried so far? -Thufir On Mon, 16 Jan 2017, Olivier wrote: > Thinking over my previous, I wonder if sipsak could be used to send > outgoing SIP NOTIFY messages. > Would both Asterisk and sipsak be able to share networks resources ? > > Thoughts ? > > 2017-01-16 14:10 GMT+01:00 Olivier

Hello, I am having a serious problem with clients that are connecting to our Mandrake 8.1 server that is using the Samba 2.2.2 RPM. After a few days of having the computers on they will start to take up 200M of ram as reported by top. I don't have this problem with any of the other clients, Win2k, ME, 98 or 95. Is there a setting I need to change on the client side? The only way to

Hello! Im trying to mount the ADMIN$ share of my windows XP and i cant! I can mount other shares. I do ./smbclient -L 192.168.0.18 added interface ip=192.168.0.1 bcast=192.168.0.255 nmask=255.255.255.0 session request to 192.168.0.18 failed (Called name not present) session request to 192 failed (Called name not present) Password: Domain=[WORKGROUP] OS=[Windows 5.1] Server=[Windows 2000 LAN

Is there a traceroute equivalent in the VoIP world? I would like to see the route a call takes after it gets to the gateway. Basically showing all the hops until it reaches it's destination or PSTN termination. -Dave

Hi, Is it possible to get the XML config off of a Linksys SPA-941 or 942 phone? I've tried http://[ip address]/admin/spacfg.xml however that file doesn't appear to exist. Thanks.