similar to: G.729 Codec on Dialup

hi, Can anybody suggest me a good, reliable, robust, SIP supported hardware IP phone? Surajee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030906/ed1d46cf/attachment.htm

hi, I have a problem in call time out, An ISDN PRI E1 from PSTN and another ISDN PRI E1 from a Nortel PBX is conneted to my server. But when i do a Dialout(from both E1s)the calls do not timeout. For ex. Dial(Zap/g4/123456|20|t) suppose other side is ringing and is not answering. even after 20 seconds, call doesn't get timeout pls gv me a solutions.. its really urgent.. Surajee

Hi Everybody I've got a weird problem. I am running one Asterisk system on a dual processor box. This box mostly do VoIP only but it has a Fritz PCI ISDN card installed with latest drivers. Dialing out through the ISDN cards from an internal Snom phone works fine and so does dialing in. Except - if I load the ztdummy module (for IAX channels) the capi drivers starts acting up. It is hard

In Sweden, where I live, it's snowing like crazy. The Stockholm area is covered in white stuff and there's really no reason to leave the computer and get out anywhere. More white stuff is coming down all the time. Boring. I am sure your weekend is no better - rain, snow or just another boring sunny day. Let's find something cool to do during this weekend! Join the cool crowd

i have been using the asterisk for some three weeks. Previously i was using the softphone iax-phone and now i have to shift to the sip phone xlite. The problem is that there's always unbearable noice in sip to sip calls. Is there any way to get rid of this???? Kindest MM Luqman -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have Asterisk running with a combination of SIP and H323 clients. I am using the OH323 module instead of the H323 one. When the SIP clients ring each other, they can hear a ringing noise in the ear peice to let them know that the other parties phone is ringing. However, when the H323 client rings a SIP client, there is no ringing sound at all, although as soon as the called party picks up the

hi All, quit new to asterisk, can anybody tell me whether Wildcard E400P and Wildcard E100P support R2 CAS protocol. if they do, what is the value, i should set to 'signalling' parameter in the zapata.conf file? Surajee -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, This question is about DBPut and DBGet, Can i put about 1000 keys in a single family, (only once for the lifetime) for ex. exten => _X.,5,DBput(family/key1=${val}) ... exten => _X.,5,DBput(family/key1000=${val}) like above and if i later retrieve it, randomely, with inbound calls, will it affect performance? Surajee -------------- next part -------------- An HTML

And someone goes and redesigns the look and operation of the Trac. Quite noice, but a couple of comments: * The main toolbar ("Wiki", "Documentation", "Timeline") ends up under the puppet logo on the left hand side, which is a pest. * I can''t click on any link, text box, or button in the main part of the page. All of the navigation links work fine, and I

hi, we are experiecing the following probem, if anybody have come across such a problem or a solution to this please let us know. our set up is, an Asterisk server equipped with, 4 port station interface card ,single port fxo card and several soft sip phones we have found problems with the following scenarios, outside caller (calling through fxo interface) <------------------------------>

Dear all I have asterisk 1.4.11 on CentOS. I have SIP IP phone arround 100 but i got Noice on voice call so what would be the resone and how to fine tune my voice quality on asterisk ?? what codec would be best for my asterisk --------------------------------- Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more! -------------- next

hi All, We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk. We were able to do one type of call transfering, ie, the called person can transfer the original call to another person. but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial application intructs to

Hello all We have tested for a mounth or two an asterisk PBX using one T1 channel bank with 24 fxs and one TDM400P digium card with 4 FXO modules. This worked with minor problems, the most notorious being some sporadic static noice or failure in the first FXO module on the wildcard. Now we have a client with 12 pstn lines and 48 extensions and we are trying to deploy an Asterisk PBX server

Hello all! First of all, this is my first post to the list. I've tried to find my answers in the forums and by Googling , but no luck. My apologies if this question has been answered before. I've set up an Asterisk box with four local SIP users. The Asterisk box uses a SIP provider for placing external calls and receiving incoming calls as well. In other words, there's no PSTN

How can I add some size to SPEEX encoded packets without affecting decoding results? I need it to fit smaller (due to VAD) packets in CBR acm-wav file. i.e. I have 10 bytes packet with 0.02s of silence (only background noice without speech). I need to fit it in 16kbps CBR wav file, so I need to put 30 additional bytes to this packets, but decoder should still decode only 0.02s of background

On 4/5/16 3:17 PM, Joshua Colp wrote: > Carlos Chavez wrote: >> I am currently having a voice quality problem with one of our Asterisk >> servers. We have checked the network and we have found no problems that >> could cause the voice to sound cracked and with small interruptions. I >> am looking at the timing source for Asterisk and it is currently using >>

Thank you, Andreas Engel. I downloaded the white paper of the Fraunhofer Acoustic Echo Control. http://www.iis.fraunhofer.de/bf/amm/download/whitepapers/Acoustic_Echo_Control-wp.pdf It said > "In the Fraunhofer Acoustic Echo Control, the frequency spectrum of the microphone signal is > modified so that the undesired echo components are removed from the signal transmitted to > the

Aaaargggg.... The S-1-5-18 AND sid S-1-5-32-544 did not resolve. But sid S-1-5-32-544 first not then later on it works.? Sorry about the noice, but that one i wanted to point out also. I hate it when im almost done with typing and mr Penny comes first. ;-) :-p Greetz, Louis > > > > > -----Oorspronkelijk bericht----- > > Van: samba [mailto:samba-bounces at

On 07/30/2013 10:03 AM, Star Tan wrote: > Hi Tobias and all Polly developers, > > I have re-evaluated the Polly compile-time performance using newest > LLVM/Polly source code. You can view the results on > http://188.40.87.11:8000 > <http://188.40.87.11:8000/db_default/v4/nts/16?compare_to=9&baseline=9&aggregation_fn=median>. > > Especially, I also evaluated

I'm wondering if anyone has seen (heard!) this before. I have a site which has Grandstream Budgetone 100 phones (don't laugh, they weren't my choice and I was quite angry when I heard they'd been installed )-: They have an asterisk box with a TDM400 card in it with 4 FXO ports and 4 lines to the telco (BT, in the UK) It basically works and does what it says it's supposed