similar to: codec translate

HI ALL; Is there anybody who use app_radius(astersik radius module)??????????? is it stable? Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040803/8a096bfe/attachment.htm

HI ALL; Anybody can explain the difference between "call parking " vs "call transfer" Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040720/2975991b/attachment.htm

HI; Thanks for your reply. The reason for why I am going through asterisk in such case is just "using asterisk voicemail service" I mean: ATA1 calls ATA2, suppose ATA2 is unreachable or he is not at the office, then the call reroute (my GK is able to reroute calls if the first route is not valid) to atersik for voicemail service. Do you think I can handle it with asterisk native

Hello, Asterisk Real time database worked on astersik 1.6.2.0 but now i am working on Asterisk to latest version which is 1.6.2.2 ,there is a a warning [Feb 24 16:26:14] WARNING[4053]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available [Feb 24 16:26:14] NOTICE[4053]: chan_sip.c:21500 handle_request_register:

HI ALL; I found in the following page: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing 1-If I could record all IVR promts in G729 format 2-If I could record voicemail in g279 format with """format_g729.c""""" then I donot need any g729 license (I suppose all my clients have g729 ip phones) My question is, how

Hi ; Have two handytone 486 and want to use them as digium TDM400 fxo-fxs card... I mean is it possible to direct pstn calls from astersik (extensions) to handytone line port directly and vice versa ?... Thanks in advance Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli

Hello list i have configured astersik 1.4 with sip i have a question when i put in dial plan.conf exten => _0678922645.,1,Set(CALLERID(number)=520460587) exten => _0678922645 .,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0)) exten => _0678922645 .,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded)) exten => _067892264*5*,2,Hangup() i can not call my

Hi all, As we know we can configure in astersik like before 5:00pm calls go to reception and after 5:00 pm calls go to some mobile no. One of my client requested that he wants to manually shift the dial plan like above as he has flexiable timing sometime he finishes at 3:00pm some time 8pm. I can not give him freepbx access. Any idea or solution. Regards Farooq --

Hi list, I'm trying to get some ideas on this subject. Normally astersik sends emails with voicemail attached trough local MTA. As far as i know there is no way for asterisk to authenticate to an external mailserver to relay these emails. Well, these days every provider has some sort of spam blocking, to add to that usually users of asterisk are behid a dynamic IP with no PTR and list grows

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031207/e5e9b8eb/attachment.htm -------------- next part -------------- Hello all, I am newbie to Telephony world (IP and PSTN). Please excuse me if you find my questions very dumb. I am trying to configure my IP 7940 with the Asterisk, when phone boots up it only shows the message "Phone

Hi All, pl help me with this basic question. I have a users (soft clients) with usernames having Alphabetics. I want to use Asterisk as my server. How should I have the dial plans as there are no numbers involved . so How can I make the configuration to work ( with numbers I can get this done using extensions.conf) my expected result is : alice at pbx.com should be able to call bob at

re-posting the question. ----------- use case: when some one in my pbx calls 100.200, I have translations well defined, Media also (media via asterisk) --Works. when some one calls bob, or for any names I am adding Domain and call is been sent to the other party -- Works, no media... For the cases when it is talking to the external work, I want Astersik not to do anything with the SDP/payload.

Hi all, I am very new with Astersik. Could some body tell me if it is possible to run Asterisk without any .conf file in /etc/asterisk ? I just want to test if my Asterisk has been installed correctly and as I am waiting for digium cards ... I have already tried but nothing happened after some verbose it stop... Thanks Angel __________________________________ Do you Yahoo!? Yahoo! Tax

Good day all I want to know what the best codec is to use for asteris for VOIP We have two towns connected with a 64k line that's going to do VOIP with astersik.At the moment with the default installation the quality is bad and the bandwith is high. Is this even a codec problem Pleas help ALtus

I have installed a 6VPCI card from voicetronix's but i can't get astersik to use it! Now looking at the loaded modules the chan_vpb is not loaded- so I assume that is why it is not working. Now I modified my vpb.conf file and extensions.conf, have I missed something Has anyone a installation guide as I am very new to this!! I have had asterisk working with SIP extensions. by dowloading

Hi; How can a user block his caller-id in Astersik? Regards Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041229/07ecf20f/attachment.htm

Hi, I try to see ExtensionStatus (event) when I'm logged on manager. But nothing :/ This is implemented in manager.c. May be I compile my astersik with out a parameter ? Someone can help me ? Thanks !

Is it possible to use openh323's t38modem with asterisk and spanddsp? or would hylafax have to be thrown into the mix? If it is possilble how would I go about getting astersik to see it?

At the most recent project I completed I have to post a intranet web page with instructions on using the system and phones. Asterisk is 1.07 stable and the phones are Polycom IP300, IP500, and IP600. Has anyone done an Astersik users guide? Something non-technical but covering most of the features an office worker would use. If nothing exists, should we develop this as a documentation project?

Hello, I am writing a program based on Astersik Manager which needs to put calls on hold and to redirect them to others extensions. I haven't funded any action able to do this. Is there a way to put calls on hold using Asterisk Manager Actions? Amaury ?