similar to: Problem with Vonage and Asterisk, Please help me

Hi All, I have installed Asterisk and able to create Users and get them connected to Asterisk after authentication. My question is how can I make calls to different DIAX clients through my Asterisk server. I also have vonage softphone account, using that I tried calling 18882255322 -- Registered 'manoj' (AUTHENTICATED) at 59.93.73.0:4569 -- Registered 'diax'

Hi All, I am a newbie to Asterisk and I was able to install Asterisk and call out. Recently I purchased two DID's, can someone please tell me or point to some links showing how to configure these DID's for SIP based softphones like Express talk? Thanks, Manoj.

Hi All, I am a newbie to VOIP and after some problems I was able to install Asterisk. If I start Asterisk I could find "Asterisk Ready" at the end and I am thinking that Asterisk is started successfully. Later after changing my Extensions.conf and ser.conf nothing works, I could still see the message "Asterisk Ready" but when I try using DIAX and connect to Asterisk nothing

Hi All, I was able to install Asterisk and make outgoing calls. Recently I purchased two DID's and I am facing a problem configuring them to my Asterisk, I hope with the help I get from this list I will be able to configure successfully. Mu errors are Feb 28 08:31:58 NOTICE[19133]: pbx.c:1331 pbx_extension_helper: Cannot find extension context 'context_mantra2' Feb 28 08:31:58

Hello *'s, First Of all Marry Christmas, I want to setup asterisk at large means "my main asterisk server placed in my office(in Pakistan), and some offices outside Pakistan and i want to connect these locations to my main * server (in Pakistan) on remote locations i'll used asterisk can i do this or may be i changed my plans kindly guides me. Thanks In Advance. Adnan Ahmed.

Hi All, I installed Asterisk recently and it was working from 2 weeks without a problem until today. Today it started showing strange error Feb 28 03:14:08 WARNING[31430]: chan_sip.c:4826 check_auth: Stale nonce received from '<sip:18006733555@mantragroup.com>' Whatever number I call it displays this, please tell how can I fix this? I have no idea what is happening and the cause

I can cut and paste the log file from a reload right now, and provide you with the other information when I get home after work: tmp*CLI> sip debug SIP Debugging Enabled tmp*CLI> reload Mar 21 14:52:42 NOTICE[23231]: indications.c:397 ast_unregister_indication_country: Removed default indication country 'us' 11 headers, 0 lines Reliably Transmitting: REGISTER

Hi All, I was able to install Asterisk and Asterisk-addons and use them successfully. But I have a problem now, I have many contexts and it looks like Asterisk is unable to find the context given directly in Mysql DB unless I specify it in Extensions.conf to switch it to RealTime. If I add a new context in Mysql then I have to add it in Extensions.conf and reload extensions whenever I need a new

Hi All, I am a newbie and trying to install Asterisk from instructions given in http://www.voip-info.org/tiki-index.php?page=Asterisk+RPM. We have Centos 3.3 so I downloaded rpm's from ftp://ftp.linuxsys.com/pub/LSE/packages/CentOS-3.4/asterisk-1.0.9/ and tried installing one by one but I get the following errors error: Failed dependencies: asterisk = v1.0.9 is needed by

mkumar@mantragroup.com wrote: > Hi All, > > I am a newbie and trying to install Asterisk from instructions given > in http://www.voip-info.org/tiki-index.php?page=Asterisk+RPM. We have > Centos 3.3 so > I downloaded rpm's from > ftp://ftp.linuxsys.com/pub/LSE/packages/CentOS-3.4/asterisk-1.0.9/ and > tried installing one by one but I get the following errors

Hi All, We are making a VOIP application for Mobiles (PDA's) and we are using Asterisk for it. We have a setup consisting of both SER and Asterisk. SER acts as a SIP router and routes everything to Asterisk. We also have rtpproxy for SER. Our packet delivery from clients (Mobiles, PDA's) is inconsistent and ranges between 10 to 60 ms delay but the average is near to 20 ms. We use SIP.

Hi All, I want to configure fax with Asterisk and I found that we can do this reliably using G711 codec only. Currently my provider is supporting G729 and G711. During the call initiation the call starts with G729 (1'st priority) and somehow if the receiver is unable to receive call then we are providing the Caller to send a fax, but at that point they are using G729 codec. At this point how

Hi All, After few problems I have installed Asterisk and changed my iax.conf. I have defined a user in iax.conf and when I try to connect that user from DIAX phone I get the following error Jan 17 23:48:16 NOTICE[16448]: chan_iax2.c:3910 register_verify: No registration for peer 'manoj' (from 59.93.66.12) In DIAX phone I gave the below to connect username = manoj password=manoj

Does anyone currently use Vonage with Asterisk? I've tried to set it up but it looks like Asterisk (at least the version that I have) does not handle well the SIP call dialog, sending a BYE with the wrong tag. As a result, when I hang up, Vonage sends back a 400 Bad Request and the call on the PSTN side does not hang up. I know that Vonage does a lot of nasty stuff which impacts UA's

Hello everyone. I've seen postings for connecting asterisk to vonage but I'm still having trouble achieving that. I have a vonage softphone and I'm trying to register to vonage using asterisk. I have not had any luck. I am behind a firewall. I've successfully gotten xlite to connect and work from the same network. When I change the port setting in [general] to 5061, I am able to

Hi, I'm trying to get Asterisk to connect to Vonage (softphone acct) to allow me to place and receive calls. I have successfully configured Asterisk to route inbound calls and send them to the correct extension, but I can't get outbound calls to work. I have Asterisk successfully registering with Vonage, but when an INVITE is sent out, I get a "404 Not Found" back from Vonage

Hi All, I tried with different configurations and referred many articles to configure Asterisk with a Vonage account I have but all my attempts failed. I am a newbie and hope this mailing list will help fixing my problem and configure Asterisk. The error I get after I make a call to outside number like 18007633555 is -- Accepting AUTHENTICATED call from 59.93.69.218, requested format =

So Asterisk can't send VOIP calls to Vonage -- but it is still possible to use Vonage for flat-rate long distance by connecting the Vonage AT-196 to an * FXS port, right? The price is an extra D/A <--> A/D conversion. Jim Shilliday IT Director Equal Justice Center 1315 Walnut St. Suite 400 Philadelphia PA 19107 215-238-6970 -----Original Message----- From: Tim Petlock

I setup a vonage account last year, and cancelled it last night when I put my asterisk box together and signed up for a Broadvoice account to use with it. Now I would like to use my Linksys router as an MTA, but realize it is still programmed with all of vonage's proprietary information and I do not know how to clear it. I understand that just pushing the reset button will not do it.

i have tried to connect my asterisk server to vonage like this: Sip.conf: register => 1<yournumber>:<secret>@atlas-east.vonage.net:5060 [vonage] type=friend username=1<yournumber> secret=<secret> host=atlas-east.vonage.net port=5060 allow=all maxexpirey=15 dtmfmode=inband fromuser=1<yournumber> fromdomain=atlas-east.vonage.net canreinvite=no nat=yes