search for: chimit

...oadRunner/Time Warner connection and able to successfully with SIP (or IAX2)? We are experiencing high latency up to the point that the voice conversation is not understandable anymore. This goes for both SIP and IAX2. Is anybody willing to share experiences or give tips? Rene Kluwen Chimit _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attach...

To All If anyone can shed any light on this it would be greatly appreciated. My phones are unable to enter pins numbers correctly when required by the party they are calling. For example I was given an outside number to attend conference bridge. After the call was connected it required me to enter a 4 digit PIN. Now here is the problem whenever I enter a pin it is received twice. For example if

Hallo all do any of you know if the toll free access to the Netherlands is still working via FWD or Iaxtel? thanks liaan --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term' -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ual format = 512 -- Called 0031651931985@mutualphone -- SIP/mutualphone-6b26 is ringing -- SIP/mutualphone-6b26 answered IAX2/iaxrene@iaxrene/2 The BT101 gives this: -- Called 003165193XXXX@mutualphone -- SIP/mutualphone-2de1 is ringing -- SIP/mutualphone-2de1 answered SIP/chimit01-6013 -- Attempting native bridge of SIP/chimit01-6013 and SIP/mutualphone-2de1 Jan 16 18:50:41 WARNING[18631600]: chan_sip.c:2804 process_sdp: No compatible codecs! -- Got SIP response 488 "Not Acceptable Here" back from 209.250.147.116 show translation (I figure this has a...

Hii, I want to forward calls from an asterisk server to a local gsm network. I have read the wiki pages on various forums. But the thing i want is to receive the call(Voip) from an asterisk server then it should be forwarded to a gsm network & again to either a gsm/ PSTN from the gsm network itself. Please post a help. Thanx in advance. -- Day by Day in Every Way I'm Getting Better

...eir claims, the problem occurs less frequently (only every so many calls). Is this a known problem with isdn4linux? Does the Linux capi driver support ISDN cards utilizing Winbond chips? Or do you guys think that changing to capi would not help me in this matter? Thanks in advance, Rene Kluwen Chimit

Does anyone know of any companies where I can interconnect with for SMS? ? .o-------------------------------------------------------o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office

Hello, I'm trying YAACID ( http://www.shatterit.com/opensource/yaacid/ ) for incomming call notification on PC (and open url with callerid), but it does not display/pop anything :-( my config is very simple... (yaacid is successfully registered as manager in asterisk) thanks PJ * dialplan: '953' => 1. NoOp(${CALLERID}) [pbx_config]

Any means of killing a .call file that is in progress? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

Hi, I currently use? Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on a debian sarge with a kernel 2.4.27 on a P4 3Gig with 1Gig of memory When i use i4l on any call, the called party ( on the telco operator side ) ear me with a delay of 1 sec after 1 minutes , 2 sec after 3 minutes and so on... After a quart hour, the delay make the conversation just

Hello all, Recently I migrated all telephony in my house to asterisk thanks to the Asterisk, QuadBRI which works wonderfully well. Some small tweaks to make but that's on the long list. On the short list is the ability to reliable send and receive SMS. For SMS I already built a script email2sms, but sometimes the SMS doesn't get send from some reason, the sms log then reports something

For canreinvite=yes to work, I think I need to remove the t argument in the Dial(SIP/ext|60|t) application. Otherwise, Asterisk will allways stay in the middle. I don't want that, so I removed the 't' argument. That works. Now, when two UA are calling, Asterisk gets out of the RTP stream. However, when removing the 't' argument, the Music On Hold doesn't work anymore

Hello, i wonder why i didn't find a solution for this problem yet, because it seems very common: I have an asterisk server with an AVM (Fritz) ISDN-Card (BRI), and some SIP-Softphones which i can call from outside by calling the phonenumber of the Asterisk-Server and then dialing the number of the SIP-Phone. If I make a call from a SIP-Phone into PSTN, only the MSN of the asterisk-server is

I bought three plus two Grandstream BudgeTone 101 phones. The shipping cost more than the phone itself from Pulver store. The first shipping had one phone defect. Nothing on the display. (Can happen!) The second shipment had one phone with a defect display, but it still worked. The second phone's handset was defect too (microphone did not work). Changing the handset from this one to the

Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if i can fill this 30 channels with REAL traffic for 2 or 3 days I can find new bugs on chan_unicall or I can see how stable it can be. Im using R2/MFC with chan_unicall the patch that Steve Underwood wrote. I will let anyone make FREE LOCAL calls to Mexico City till saturday or maybe until monday to see how stable this can

I have an * system running version 1.0.8 and it works mostly fine. I am using it as a virtual PBX and we share the box among companies. I have the calls all staying separate, we well as the companies' extensions, voicemail, etc. The only problem I'm having is with two accounts that use the same SIP termination provider. * seems to be confusing the sip contexts for the incoming calls.

Hi, I've been trying to dial out to a cellphone, but all my calls get redirected to 066 (the emergency number at my city, like 911) does anyone know how to fix this, any ideas,? does anyone from mexico has done this? Any comment will be highly appreciated, Regards, Claudio -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I've signed up for a NuFone account, and added the following instructions to my config files per NufFones directinos: iax.conf [NuFone] type=peer host=switch-1.nufone.net secret=password extensions.conf (under the [default] context) exten => _1NXXNXXXXXX,1,Dial,IAX2/f00b3r@NuFone/${EXTEN} I then get this message in the CLI: -- Executing Dial("SIP/jake-fe5d",

Hi, Here's the deal. When someone leaves me a voicemail message I want Asterisk to call me on my cellphone by dialing my cellphone number and tell me I have a message. Is this possible? Can anyone cite examples? Most commercial voicemail systems produced in the last 10 years can do this. Any help would be much appreciated. Regards, Mike -------------- next part -------------- An HTML

I have upgraded the GXP-2000 to the newest firmware 1.0.1.12<http://1.0.1.12>and the phone is much more usable However, I still have two slight sound quality issues: 1) There is static on the line at all times. It is not that noticable to me, but when I make calls out the PSTN the person on the other end hears it. If I use a Cisco ATA with an analog phone and call the same person again