similar to: Converting .wav to .WAV

Hi, I'm trying to edit an AMP-derived dialplan: the macro "dial" uses the AGI script "dialparties.agi" to find the extension to call. I'd like to drop this script: does anyone can explain me what is its main job? Thanks -- Domenico Viggiani

Hi, I'm planning to install a new Asterisk server with a Digium TE410P card. Can I use Red Hat Advanced Server 4 (latest update)? Is this a good choice? Is recompiling Asterisk simple with kernel 2.6? Thanks -- Domenico Viggiani

Hi, I'm trying my new Snom 360 phone (6.2 firmware) and I'm seeing that it doesn't register with the Asterisk 1.2.9.1 server after a reboot. I need to click "Re-register" in the web interface. I set: - Support broken Registrar: On - RTP Encryption: Off Any help? -- Domenico Viggiani

Hi, during gradual migration to Asterisk, I put Asterisk in front of a legacy Alcatel PBX: PRI PSTN <--> Asterisk <--> E1 cable <--> Alcatel PBX After successful deployment of VoIP phones, it's time to drop Alcatel PBX! I'd like to keep some of analog lines to support modem, fax and some older stuff. What's the best choice? A channel bank or a TDM2400P card? Can I

Hi, is there a RHEL4 RPM for the Festival text-to-speech system? Thanks -- Domenico Viggiani

Hi, calling a partner on the other side of a SIP trunk, call gets disconnected immediately after answer. This is the content of log file: Jun 14 16:25:14 DEBUG[14380] channel.c: Didn't get a frame from channel: SIP/cerved-out-6eba Jun 14 16:25:14 DEBUG[14380] channel.c: Bridge stops bridging channels SIP/232-2e41 and SIP/cerved-out-6eba Jun 14 16:25:14 DEBUG[14380] channel.c: Hanging up

Hi, I have a lot of GXP-2000 phones not registering with Asterisk server. After two days of attempts, it seems that problem is due to the fact that phones and server are not on the sme network. Do you know if this is known issue? -- Domenico Viggiani

Hi, I have a IAX2 trunk between two sites (connected with an high bandwidth link) but sometime/often I get: chan_iax2.c: Auto-congesting call due to slow response and call is dropped (and routed on a PSTN link). In iax.conf, I have: [iax-out] username=iax-in type=peer trunk=yes secret=xxxxxxxxxxx qualify=yes host=xxx.yyy.zzz.32 auth=md5 Any idea? Perpaphs is due to

Hi, I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (only one until now!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR

Hi, during tests, I configured different SIP accounts on the same phone. Now I see this 'sip show peers output': Name/username Host Dyn Nat ACL Port Status 259/259 10.97.1.19 D 5060 OK (8 ms) 232/232 10.97.1.19 D 5060 OK (7 ms) where both extensions are registered and have the same IP. But now I have only one extension

Hi, I'm actually using a slightly old version of AAH with Asterisk 1.2.1, because at first install it was perfect for my moderate knowledge of Asterisk. It is working well but I gradually introduced many changes to dialplan during normal use and now I'm feeling like in a straitjacket! Moreover I'd like to have the chance to upgrade Asterisk regularly. I have not the experience to

Can someone explain me this message: "chan_iax2.c: Ooh, voice format changed to ..." Where can I find a list of numeric codes used to identify voice format? Then, sometime I get an infinite loop of messages like these: DEBUG[15015] chan_iax2.c: Ooh, voice format changed to 1 WARNING[15015] channel.c: Unable to find a codec translation path from g723 to alaw DEBUG[15015]

Greetings all - I'm interested in using an asterisk box to supplement and add VoIP capabilities to our legacy InterTel Axxess PBX. After searching through the list archives and through google, it seems that the best way to go about this is to connect the two systems via a T1. Is this correct? The PBX currently doesn't have any VoIP capabilities, so that's not an option for

By the way, I was delighted to find out that I could join .ogg files with cat, i.e. cat {file1.ogg} file2.ogg > outputfile.ogg <p>I successfully used this capability, together with oggenc and sox, to join two .WAV files in the following manner: <p>oggenc -q9 file1.wav ... oggenc -q9 file2.wav --- cat file1.ogg file2.ogg > outfile.ogg ox outfile.ogg outfile.wav I get

After somebody records a message asterisk notifies me and encloses the WAV file. Though I'm not sure if this is a WAV format. I can not play it. According to the file specification it is: msg0000.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz How to play received message? -- #Joseph

Hi, I'm trying to write a small Visual Basic app to throw a popup with CallerIDNum when a call center agent answers a queue call. Does anyone know what is the right manager event to intercept? Thanks Mimmus

I already posted this message but I had no answer. Sincerely, I think it is a bug and I'd like to hear developers on this. Platform: - HP-UX 11.00 - HP C/ANSI C Compiler (B.11.01.06) Copying a (large) directory structure from a NT share, interactive command: # smbclient //machine/share password >prompt >recurse >mget * fails to copy 76th, 115th, 154th file of

I'm using a few GXP-2000 with firmware 1.0.2.13 and everything seems to be working fine. However, there are a couple of issues I'd like to know if are possible: 1) Even though the phone has 4 line appearances, if I am speaking on a line, the phone can no longer receive phone calls. I can manually select another line and make calls, but when Asterisk tries to send a call to it, I

Okay, before I submit my patch to make libao produce sorta-streamable wav files, I want to know what these partial wav files do to various players. I've posted two sample wav files on my webserver: http://volsung.dhcp.asu.edu/~stan/infinite.wav http://volsung.dhcp.asu.edu/~stan/zero.wav The first uses a riff and data length of 0xFFFFFFFF (approximately infinite) and second uses a riff and

Hi all, I have to play a alaw file with .wav ext. How can I do this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100721/de46328f/attachment.htm