similar to: voicemail scenario

Dears my Internet Provider , prevents , sip connections, between sip client(sip phone) and sip server, (asterisk + ser) . both of client and server are mine. is there any solution for tunneling the sip packets? best Mani ____________________________________________________________________________________ Don't pick lemons. See all the new 2007 cars at Yahoo! Autos.

hey list, we are currently migrating our users from novell to samba. now we have one problem: in novell we could give e.g. user1 access to users2 home share so he could modify, delete or add files on this share. in samba we defined a global homes share that is mapped on logon. so how can we give user1 the needed rights? here is the definition of the homes share in smb.conf: [homes]

I would like to determine through an AGI script if a specific SIP extension has DND on or off. I know that if the SIP client dialed *78 or *79 it is usually enough to just do a: database show dnd to fetch the DND status from the database. However, not all clients dial *78 or *79 (or whichever feature code is defined for DND). Some softphones such as SJPhone have a DND button. When pressed and

Dear all I have one more question about codec what codec i use for LAN setup G.729 or Alaw which is best for LAN setup caz some people told me G.729 is use for wan link not for lan caz it is cost effective so can anyone suggest me best codec for asterisk and SIP phone Rgds satish patel --------------------------------- Don't pick lemons. See all the new 2007 cars at

I''ve always wanted to be able to do stuff like this in my functional tests c = customers(:customer_1) c.expects(:great_customer_service) post :service_customer, :id => c.id This of course fails because inside the rails action a different instance of customer is used. Some of the time setting your expectation/stubbing on Customer.any_instance works, but it''s not

Hi - I am currently developing a RTSP/RTP/SDP solution to stream Speex encoded data. Using my current source, I have successfully streamed u-law and PCM encoded audio but have been unsuccessful thus far with Speex. Because of some constraints of my system, I am encoding audio at 11.025kHz. I am still using the 160 samples per frame which makes my frame size 28 bytes. I have successfully

> I believe there is new resampling functionality in the speex svn head, > although I haven't tested it yet. You might also want to check out > 'Secret Rabbit Code' for your resampling. Yes, I've just been working on a resampler recently. Its changing a lot, but it's now usable. I'd actually be quite happy to have some feedback on it. Jean-Marc > Hope this

Thank you very much. For some reason "emailsubject" was not included in my example config. Well, it's working great now. Last question, I promise :P. Is it possible to change the date format? I want it in Norwegian. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Barry Flanagan Sendt: 22.

Hi all, I''ve been following the puppet project for a while now and I think it''s starting to become mature enough to be used in a pilot project at my corp. I have a few questions: 1. What is the current largest known number of Puppet managed hosts? 2. What are the average network demands on a single puppet master? 3. Has Oracle ever been managed by Puppet? 3a. Has Oracle ever

Hi I have a problem w/ voicemail, the subject message is corruption when used voicemail variables, e.g. : voicemail.conf emailsubject=${VM_MAILBOX}|${VM_MSGNUM}|${VM_CALLERID}|${VM_DUR} Return: Subject: =?utf-8?Q?1504|12|=22Teste_-_Rafael=22_=3C1570=3E|0=3A16?= Expected: Subject: 1504|12|"Teste - Rafael" <1570>|16 Thank's Att, *Rafael dos Santos Saraiva* Tel: (51)

Greetings; I've got a situation where I really need to be able to select from a number of various (body) textx for email/pager messages when voivemails are left, but can't seem to figure out how to change them on the fly. Now, I know there are pre-defined defaults, which can be overridden by entries in the [general] section of voicemail.conf, but that does not get me there. I need to be

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

I would like to know if atxfer is supported somehow because there seems to be little documentation for this feature. I know most people expect a good SIP/IAX phone to do the job but I think it's nice to be able to do attended trasnfers with a simple ATA-connected analog phone. I have Asterisk 1.2/Freepbx and features.conf has a line regarding atxfer and I set it to *2 (Default). While # works

Hi, I am running nut with megatec driver accessing ttyS0 as user nut on "standard" kernel (gentoo-sources). It works fine. However, I just built a hardened kernel on a new gentoo machine and have no experience with it. NUT (upsdrv) is failing because it says it doesn't have permission to access ttyS0 even though nut is within the appropriate group. I can add user = root in ups.conf

Hi, I want to transfer the call to a conferencing room while dialing. I tried to do that using manager API(Redirect), but it did't work. Regards, Jason. ____________________________________________________________________________________ Don't pick lemons. See all the new 2007 cars at Yahoo! Autos. http://autos.yahoo.com/new_cars.html

Hi List; I think my problem in Zaptel compilation is related to autoconf: no input file, anyone has an advise? Also, I did a change in the Makefile existed in the following path: /usr/src/kernels/2.6.20-1.2319.fc5-i686/ EXTRAVERSION = 2.6.20-1.2319.fc5 Now, if I run uname -r then I get output: 2.6.20-1.2319.fc5 But the directory under the kernels is: 2.6.20-1.2319.fc5-i686 So do I have to

It's fixed now. In "/etc/mail/ssmtp.conf", this ("FromLineOverride=YES") line was commented out. Removing that comment did the trick :) Now I only need to change the e-mail's title. Is that possible? -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Barry Flanagan Sendt: 22.

Hi all, I use this command to see disk usage: du -cks * | sort -rn | while read size fname; do for unit in k M G T P E Z Y; do if [ $size -lt 1024 ]; then echo -e "${size}${unit}\t${fname}"; break; fi; size=$((size/1024)); done; done The data itself is around 200GB with lots of subdir and small files. The command takes for about 1 hour to finish. Is there a faster way to see the disk

Hi, I've been trying to use the FLAC lib for c++. I need to implement basic encoder function in my application, so i tried to include encoder.h , set everything to the default path, and included stream_encoder.cpp,stream encoder.cpp files in the project. Include is #include "FLAC++/encoder.h" i write something like this FLAC::Encoder::File test; i build, compile goes ok but then

Hi, This word for a little problem : when using the Equalizer integrated with Winamp v 5.24, this has no action on FLAC files! With MP3 files, no problem, and when reading FLAC files using the Creative Equalizer, no problems too! Is this a bug in Winamp ? Thanks, Didier Didier Eggerickx Rue de Cambron, 48 7870 Cambron St Vincent GSM : 0484/730491 E-Mail : didier.eggerickx@skynet.be