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...sk using the same IP address. Now, wenn all are registered, all the calls get routed to the client that registered most recently, but not to the correct client. Also, there seems to be some problems registering all 3 clients simultaneously. Could anyone help, please? Thank you very much, Roman Zhovtulya

...recently, but not to the correct client. Also, there seems to be some problems registering all 3 clients simultaneously. I have NAT=yes in all users and "nat=no" is commented out in sip.conf Is there any other place to check? Could anyone help, please? Thank you very much, Roman Zhovtulya

Dear all, I've noticed some significant voice quality deterioration when calling US landline via VoIPjet.com in the last week or so. Before that the quality was pretty good. Has anyone else experienced any voice quality problems with voipjet recently? Thanks, Roman

...I execute this, it says that "User entered ''". Why wouldn't it read the numbers punched on the phone? The Voicemail works very well. I use dtmfmode = rfc2833 and iLBC codec. Also, please check if the comments I made to the code below are correct. Thank you very much, Roman Zhovtulya

...version. Isn't there an easy way to redirect a call coming from a provider (sipgate.de) to one of the extensions based on the last 3 numbers a caller entered? By the way, I'm using RealTime (mysql) for sip (just users) and extensions. Can it pose a problem? Thank you very much, Roman Zhovtulya -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Adam Goryachev Sent: Donnerstag, 3. M?rz 2005 15:05 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] (another try) Dial...

hello i am using asterisk-oh323-0.7.1. i want to convert sip call to h323 (h323 sjphone or h323 proxy). what could be the best way for this. i am successfull in converting h323->sip by using asterisk as gateway. help required on sip->h323. kamran __________________________________ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web

Hi all, I registered a SIP account at budgetphone.nl/talkin2ya.nl Receiving calls works like a charm, I even redirected my normal PSTN number to the number I got from them so everything ends up in my * server. Before I ask them to take over my normal phone number I wanted to test all of it, so I ordered some calling minutes to test. Now I cannot get outbound calling to work with them. Anyone here

I have been searching all over for the answer on all sources online and have come to the conclusion that it must be rudimentary or I am asking the wrong question. I cannot figure out how to configure the box to set the "from" address to a correct domain, as my outgoing isp will not pass mail from root@asterisk1.local, as I expect it wouldn't. Any help is appreciated, even just

I am setting up a new asterisk based call center. I just read: http://www.voip-info.org/wiki-IAX+versus+SIP After reading this and other google results for "IAX vs SIP" is there any reason why i should use SIP anywhere !! t

Hello, I was trying to configure 2 Digium X100P-Card compatible FXO cards. Everything went OK up to starting Asterisk, the following error occures while parsing zapata.conf: Parsing '/etc/asterisk/zapata.conf': found Mar 27 16:24:20 ERROR[1075656544]: chan_zap.c:5196 mkintf: Unable to get parameters Mar 27 16:24:20 ERROR[1075656544]: chan_zap.c:7162 setup_zap: Unable to register channel

Hello, I'm working with one institution that considers deploying Asterisk as their internal PBX. To overcome their initial "undecisiveness", I would very much like to present them with some examples of Asterisk deployment in similar environments. I'd be happy to have some info on: Asterisk installations for 100-500 users in company/government environments Deployment reports