similar to: Bug in SMBCLIENT

I'm posting this message in charge of Denis MARS (France), because he is not a subscriber of teh list and I worked with him about the same problem. Domenico Viggiani ---------------------------------------------------------------- Hello, I worked on this problem since three weeks and now i really give up. Clearly i can't go further without your help. I hesitate to send you this bug

Hi, how can I convert .wav files to .WAV: # file greet.* greet.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz greet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz using 'sox'? 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 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'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 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'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

Quick question -- if I have a vector of strings that I'd like to split into two new vectors based on a substring that is inside of each string, what is the most efficient way to do this? The substring that I want to split on is multiple characters, if that matters, and it is contained in every element of the character vector. --j -- Jonathan A. Greenberg, PhD Postdoctoral Scholar

On Jul 22, 2013, at 12:56 AM, Duncan Sands <baldrick at free.fr> wrote: > my understanding is different. I'm pretty sure that what I'm about to say is > the traditional way these things have been viewed in LLVM. That doesn't mean > that it's the best way to view these things. > >> - nounwind means no dwarf EH. Absence > > I guess you mean

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, 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 Andrew, On 22/07/13 10:23, Andrew Trick wrote: > > On Jul 22, 2013, at 12:56 AM, Duncan Sands <baldrick at free.fr > <mailto:baldrick at free.fr>> wrote: > >> my understanding is different. I'm pretty sure that what I'm about to say is >> the traditional way these things have been viewed in LLVM. That doesn't mean >> that it's the

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

New to this list. Running samba 2.2.4 set up as a PDC. We've recently added another department to our office network. Ideally, I don't want either department to see each others shares - just to be aware of the shares in their own workgroups. The documentation says that you can set up virtual servers using the "netbios aliases" and the "%L" options in the

Does anyone know if the Marriott has Wi-Fi? LAN connection in the room? Mike

Good afternoon, Hope you all have a wonderful day. I am glad to be here. Hope you could help me with the following errors that i have been trying to figure it all out since last week. I am using Splus from Insightful, and as i read, R and Splus are very similar. So hope you could help me. I have been continously received these error messages after i ran my small program for couple of

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

Hi there, I'm getting a bunch of these errors from Polycom phones in 1.2.1: ERROR[24301]: chan_sip.c:10790 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 4003 in context internal I've searched the Wiki and archives to no avail - what do these errors mean? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web:

I'm pleased to announce that Flash Operator Panel 0.26 has been released! FOP is a GPL'd switchboard type application for the Asterisk PBX. It runs on a web browser with the flash plugin. It is able to display information about your Asterisk box in real time. It is included in FreePBX, Asterisk@Home, DeStar, startShop, and several other projects both free and commercial. You can grab the