similar to: transforming g729 files to wav files

Hello, How to reproduce this bug (?) : 1. register sipura spa2 with 2 lines on asterisk. 2. use first line to call somewhere. 3. while using first line try to call from second line somewhere else in 3 step i hear fast busy tones on second line and asterisk console gives me this short error: Mar 27 07:01:43 NOTICE[29656]: chan_sip.c:3629 process_sdp: No compatible codecs! My sipura adapter

Hello folks, firstly, thank you for your useful and fast answers ! Is there anybody using D-Link SIP phones ? Are D-Link SIP phones ok to install in production environment ? Give your comments please. Tofik Suleymanov

Hello list, When new voicemail comes and i pick up the phone i hear special tones indicating that the new voicemail arrived.I've never had any problems with this feature, but several days ago it begin to behave strangely: 1. new voimcemail arrives, but i dont hear the special indicating tones when picking up the phone 2. there is no new voicemail (checked mailbox on filesystem), but when i

Did anybody know, Is it possible to establish a ISDN DIAL up Connection and Analog Dial up Connection (V90) trough asterisk with Digium TE405? Thanks a lot for help. Nico Giefing -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060328/3b657058/attachment.htm

Hi to all, I already setup asterisk on REDhat 9.0 linux machine. I will have 4 physical phone lines and 10 IP phones for it to use. I have a network setup already. Is getting TDM400P - 4port FXO from digium enough to start? Do I need anything else? Thank you

Hi, I've tried the link http://www.asteriskguru.com/tools/audio_conversion.php but it returns an error at the moment. Any other ideas most welcome. Tx Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090423/c491a7b9/attachment.htm

This is my second attempt trying to get help and I am hoping someone can. When the musiconhold extension is matched, Asterisk attempts to execute musiconhold and stops right away, this is what I gets: Executing MusicOnHold("OSS/dsp", "") in new stack -- Started music on hold, class 'default', on OSS/dsp -- Stopped music on hold on OSS/dsp Is there a file that

Hello Asterisk, noticed that the mbrola really adds a extra dimension to tts; anyone got any experience with running this together with the festival and * ? -- Best regards, Danny mailto:dannyz@belgonet.com belGOnet.com a Euro-pictures division - internet solutions place princesse elisabeth 9/11 - 1030 Brussels - Belgium Tel : +32-(0)2-215.67.65 -

Hello, I try configuring Quintum DX gateway as link to PSTN for *. Now, I can dial number which is connect to Quintum, and call is diverted to *. I don't know what I should set, if I want call from SIP_phone registred in Asterisk to PSTN via Quitnum. I set in sip.conf account for Quintum [sip_proxy-out] type=peer outboundproxy=QUINTUM_IP , and changed extensions.conf. When

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Hello, I got Quintum A800 with the P5-2-1 firmware. I configure my asterisk trunk as followed: [SIP_BD1] type=peer qualify=yes host=192.168.0.254 disallow=all context=from-pstn allow=h723 and inside the quantum I change the config sip useragent to 5060. Up to this part if I run sip show peers, I got: asterisk1*CLI> sip show peers Name/username????????????? Host??????????? Dyn Nat ACL

Hi all, Can someone share with me his experience in making asterisk-oh323 talk to quintum gateway without gatekeeper. My set up is QUINTUM GATEWAY ------IP----M ASTERISK (OH323) Both are gateways.. but I don't know what authentication I will set up in oh323.conf and how to set it up I will be glad if anyone can help Goksie

Hi all! I'm getting an error when I try to start asterisk with chan_misdn. I patched my kernel (2.6.12.4), and compiled the whole stuff (kernel, mISDNuser, asterisk, chan_misdn). I got mISDN from http://isdn.jolly.de/download/v3.0/ I'm using a CVS Snapshot of asterisk, which was checked out about 5 hours ago. This is the error: [chan_misdn.so]Aug 15 14:13:29 WARNING[4929]:

Hi all, -------- beacause I am a newbie in the asterisk ralm and the existing documentation could not satisfy I'd like to ask you some Questions: 1. Does somewhere in the Internet exist additional documentations for asterisk configuration ? 2. Does Asterisk work as a standard SIP Proxy ? 3. I am just installing a Asterisk PBX in our institute and additionally I purchased some ot the Snom

Hi, I'm having problems to register Quintum Tenor DX 3020 on a Asterisk box with SIP. Asterisk always returns "Username/Password mismatch". I've tried all configurations that was on the Quintum's manual, but no success. I've tested the same username and password with a Linksys (PAP2-NA) with the same asterisk box, and it worked fine. Where is the problem ?

Tofik Suleymanov wrote: > Steve Totaro wrote: > >> Give idefisk a try. It works very well for me, its free, and does >> not crash all the time like Cubix (formerly Firefly). >> >> > > Hello Steve, > > As far as i know 'idefisk' is a softphone, but i need a hardware phone. > Thank you for reply. > > Tofik Suleymanov > Oooops, sorry its

Hello All, I am trying to setup a small system where Nextone Softswitch will send traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog Gateway but for some odd reasons the call are flashed back from Grandstream to Asterisk and creating a Black loop... I did follow the instructions provided by Grandstream support but it doesn't seems to be working...

Hi All I'm trying with no luck to connected the Quintum D series Gateway with the new SIP release to asterisk. Have anyone done this? If yes then how should I configure the sip.conf to accept the registration? maybe a sample config? Thanks /Krystian

Hi! I'm goining to connect an AVM C4 to 4 BRI-lines in PTP-Mode, all BRI-lines will have the same number (1234-XX for example). I don't know if this is possible and I can't test it, because I don't have these lines yet. Is anybody out there, running such a configuration? I'm looking for some example configs. Especially I want to know how it is possible, to let asterisk

Hello all, I post this issue thinking too that could help other people on an asterisk deployment over distributed offices considering both quality, prices, devices and so. Well, i am working on a deployment of a telephony system based in asterisk. My company have a central office with seven remote offices connected all through a VPN. To reduce and evaluate costs i consider solutions like: