similar to: Codec conversion sip peer <> Asterisk

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 tried to use SIPSAK to understand the troubles i'm having about sending my voice to the person I've called (extension), after doing this tests below I always got this error "error: this FQDN or IP is not valid: voicegw". This could cause problems (namely audio problems)? Best regards, Helder voicegw:~# sipsak -C empty -a password -s

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

Hi! Is there a way to avoid being "at the middle" of communications between two SIP endpoints? So that we can avoid loosing bandwidth with it? Is there a way to "forward" the authentication to a IAX provider and "transfer" the call to it, avoiding using my own bandwidth? I've tested it with SER with some results, I was wondering if it is possible with

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! Could someone give me a hand? If I dial 200 for echo testing it works... Everytime I dial an extension ex. 505 get the error below.... In this example it was from 508>505 a Xlite Pro to a TA. I believe it has something to do with the way i'm executing the command dial but I use the "standart" that comes in the samples from asterisk. *CLI> -- Executing

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: >

Hi, Anybody out there running Adit600s? I have in my care an Adit600 channel bank connected to an old (version 1.0.6) Asterisk instance with MGCP. When trying a more recent Asterisk (1.4.21.2~dfsg-3+lenny1, Stock current Debian) calls fail. I have attempted to add the "slowsequence = yes" line to mgcp.conf. (It seemed to be the only likely candidate in the example files I found

Is there anything I can do regarding the linker issue? The solaris linker is in /usr/ccs/bin/ld but I think llvm wants to use the gnu linker. gcc in my system was compiled with the solaris linker. Thanks, Jorge On Mon, Jun 24, 2013 at 6:18 PM, Stefan Teleman <stefan.teleman at gmail.com>wrote: > On Mon, Jun 24, 2013 at 6:17 PM, Jorge Rodrigues <skeept at gmail.com> wrote: >

Hi, I am running FC1 with latest patches of 3/30/04, and I have the latest CVS as of this morning 3/30/04 of asterisk, zap and libpri. The SIP device I am using is a Sipura SPA-2000 with G726-32 "Forced". When I 'make clean" and recompiled zaptel, libpri, asterisk and start asterisk I can see: [format_g726.so] [format_g726.so] => (Raw G.726 (16/24/32/40kbps) data) ==

To me it's a error if I can't complete calls using the ATA configured to use the g726 codec. I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I received NOTICES and WARNINGS, but I can't complete a call. On a zap channel: -- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack -- Called 1/2217008 -- Zap/1-1 answered

Hello list, Grandstream GXP2010 phone calling to Snom 320, Asterisk in the middle. Grandstream allows for 8 different codec specifications. I have defined them as 4 x G726 & 4 x alaw. Snom allow for 7 different codec specifications. I have defined them as 3 x G726 & 4 x G729. The SIP peers are both defined as : disallow=all allow=g726 allow=alaw allow=g729 allow=gsm This is the

Ok, With everything restore on rtp.c, still I have no audio however the call is not destroyed immediately as before. I'm going to put a second Granstream box, and findout if between two boxes this happen too. I cannot believe that we cannot do 2 g726 on the same box at one time. Carlos -----Original Message----- From: Carlos Alperin [mailto:calperin@senecacom.net] Sent: Wednesday,

Hello, I'm wondering if anyone has benn able to successfully get g726-16 passthrouhg to work? I am wanting to use this codec instead of g729 as I'm running out of DSPs using a high complexity codec on the Ciscos. I would think it would work just as g729 does, which has been working fine for me, but it does not. G726-32 does work great however, but it's like Asterisk doesn't

Dear all, I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able to, I got the following error message: Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with our capability 0xfe02. I do not understand why because my Asterisk box load these codecs properly! Does somebody

Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd like to play PCM WAV files instead of eg. GSM. Per... www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk ... I recorded a sample of my voice using XP's Sound Recorder, then ran the following : sox test_wav.wav -r

With sipura (I tried this with both the 3000 and 841) set to prefer the g726-32 codec, a call from the sipura to asterisk will use g726. Asterisk sip.conf has: disallow=all allow=g726 allow=gsm allow=alaw When the call is from asterisk to the sipura, asterisk will not use g726. It ends up using alaw. I usually use stable but I tried this with head too, and same thing happens. Anybody know how

I have recently started experimenting with Asterisk. I am running the system the other side of the a NAT router and trying to connect to FWD. I have opened UDP ports and have configured sip.conf to handle NAT. The problem: I can call from the FWD phone and the extension on Asterisk rings and there is two way sound so no problem. Now if in the extension.conf file I have, exten =>

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