similar to: callparking vs calltransfer

HI ALL; Is astersik enable to translate between different codecs. I have couple of SIP-UA , one with (a-law) and the other with (g729), registered with my astersik box.Can astersik translate between alaw-g729 and vice varsa. Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 list, when I put a call in parking and take it back, I'm not able to put it again in parking. Context is empty and I receive message that extension 7 (or 70 if I'm quick) is not existing. Is this a bug or misconfiguration? Cheers -- Daniel

Hi List; [incoming] include => parkedcalls exten=103,1,Dial(SIP/Bob,,tT) exten=104,1,Dial(SIP/Charlie,,tT) When we use tT and when we use t alone or T alone, I know this for call parking, but I do not know what the tT does? Regards Bilal ____________________________________________________________________________________ Sucker-punch spam with award-winning protection. Try the free

Hi I was wondering if there are any special settings that I need to be able to transfer calls. Whenever I press the 'recall' button, I just here a click, and no ring-tone to transfer. in my debug log I get this : -------------------------- Feb 24 09:09:27 DEBUG[19216]: Exception on 10, channel 1 Feb 24 09:09:27 DEBUG[19216]: Got event Pulse Start(14) on channel 1 (index 0) Feb 24

HI All; I want to use Asterisk for just Voicemail Server and I need a Dynamic creation of Mailboxes. My users 's Mailboxes are same as "Extensions" but I donot want to add mailboxes in "Voicemail.conf" Is there any way to create mailbox from Asterisk dial-plan ? Appreciate any suggestions Mohammad Mirzaee -------------- next part -------------- An HTML attachment was

Dear All I want to use asterisk for the following Senario and Need help to find a SAMPLE extension.conf Incoming call >>>>>>>>>>>Asterisk >>>>>>>>>>>>>>GSM Termination Gw first leg second leg What I want to do is putting first call leg on

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; I have problem converting a windows .wav file to .gsm format by Sox. Could anyone help. Cheers, Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050706/3408bfd5/attachment.htm

Hi, I wonder if it is possible to perform the "find_by_contents" on a subset of fields indexed in acts_as_ferret.If so, how? In my code I have: acts_as_ferret (:fields => [''title'', ''focus'', ''purpose'']) However, I like to have two search options one on all fields and one only on the title. Any help is most appreciated.

HI ALL HI MICHAEL; My name is mohammad and I am iranian.I have been trying to install oh323 channel but I come up with dead end. In fact it makes me crazy. plz help me michael. I saw mailing list and I trid serevel CVS headers such as , 2004-06-07( seven of june) 0r 2004-07-02( second of july) besides I use: 1-openh323 v1.12.2 2-pwlib v1.5.2 3- asterisk CVS (2004-06-07,

Hi; Any idea of how to have different ringing tone on called party for different caller-id by means of "Alert-Info" header. Regards Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041226/3611eeae/attachment.htm

Hi; I use the following asterisk, openh323, pwlib: asterisk = cvs-head-03/09/05 openh323 = 1.13.5 pwlib = 1.6.6 asterisk-oh323= 0.7.1 Asterisk, openh323, pwlib were compiled successfully but when I try to compile Asterisk-Oh323-0.7.1 , I got the following error: chan_oh323.o chan_oh323.c chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory ......... ...........

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

Hi list, I need to represent a relationship between three tables: Tags (id, name) Users (id, name, email, ... ) Documents (id, title, ... ) I created a forth table called Assignments(id, tag-id, user-id, document-id, date). I have couple of questions: Should I use belongs-to and has-many to capture this? If so, How can I do that? should I have the id as the primary key in Assignment table or

Hi All - Anyone know if the "callpark" feature is in ABE? Is there a comprehensive list of the differences between ABE and the open source version? I've only seen a bullet-point chart which has no real detail. Thanks, Noah

Hello, This morning, I asked myself if WebRTC could be a viable alternative to softphone deployment. For me, main issue with Softphones is the amount of work needed for installation and configuration. Also, Softphones must be carefully choosen if Deskphone-like quality is expected. Now that WebRTC becomes ubiquitous, it might make sense to trade Softphone features (call history, BLF, ...) for

On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez <cursor at telecomab.mx> wrote: > > On 9/26/2018 4:46 AM, Olivier wrote: > > > Hello, > > > > This morning, I asked myself if WebRTC could be a viable alternative > > to softphone deployment. > > > > For me, main issue with Softphones is the amount of work needed for > > installation and

I'm wondering whether there's a problem with the blindxfer and atxfer commands. I was using Asterisk STABLE and pressing the # key to transfer calls worked fine, except of course when you called up FedEx and they asked "Enter the number of packages, followed by the Pound key". I found on the wiki (http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf) that

Hey all, Wondering if this is possible.. Incoming call is answered through X100P, then an extension is dialed using the same X100P card. Basically I want to dial in, enter 9 + <phone#> and have it do a flash then have it dial *08 <the same phone number> + # on the same PSTN line to have it transfer my call to another phone number. I realize this isn't very safe, but I would