similar to: SIP auth in/outside of nat?

Hello, Please forgive me if this has been addressed previously. I have been searching the archives and have not come across what I thought was a solution. My * server is behind a DSL router using a NAT IP address of 10.0.0.9. A colleague running XP and X-Lite can register with * from his home, specifying my public IP as the SIP proxy in X-Lite (however this is only true if I have the NAT flag

Hi, I am can make local extension to and from SIP X-Lite softphone, but I can't dial out using X-Lite but local analog works just fine. Here are my conf files any idea? Thanks, David my sip.conf [general] bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) allow=all [3000]

Hi, I am can make local extension to and from SIP X-Lite softphone, but I can't dial out using X-Lite but local analog works just fine. Here are my conf files any idea? Thanks, David my sip.conf [general] bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) allow=all [3000]

Hi guies, I know I know this subject have been The most written subject about VoIP Right... but I just want to make clear, just one time ! If Asterisk is on a Public IP Address and a softphone behind the nat, sip.conf must contains for this phone: nat=yes .... Now if I want to configure my sipphone (X-Lite) placing behing the NAT, it must have in "Domain/Realm" the external IP

Hello; I have asterisk?Asterisk 1.8.23.0-vici and Polycom 331 and I am able to register from local area network and not able to register from outside the office. Also from outside the office, I am able to register via PhonerLite softphone and not able to register via Zoiper softphone. So from outside the office, I am not able to register from Zoiper softphone and not able to register from

Mateo, Dialing the extension to your softphone is the same as any hardware extension. Exten => 1000,1,Dial,(SIP/1000,20,trf) pretty exten => 1000,2,Macro(vmessage,1000) exten => 1000,3,Hangup Change [mateo01] to [1000] in your sip and you will be saying that ext. 1000 is registered with the specifics you are using. Update the settings in your softphone to register the name and

We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. So far, we've been able to get the phone to register with the asterisk server, and it can receive voice from the asterisk server (IE, voicemail, etc). However, asterisk can't hear anything from the softphone. We have used 2 different machines to test this on. We are watching

Try the snom soft phone! http://snom.com CS > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Dave Chase > Sent: Saturday, February 26, 2005 12:31 PM > To: ich@mateo.ch; Asterisk Users Mailing List - > Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Anybody using

Hello All, I have a number of X-Lite users in countries where the incumbent Telco will do anything to block VOIP traffic. For some reason neither the X-Lite broadband or dialup clients would register with my server unless we configure them to use the Xten Xtunnels demo server. Once the Client has registered the call quality is great! The problem is the Xtunnels does not support other IP Hard

Good day, I have a puzzling issue that people in the IRC channel recommended I post to the list so here goes :) I am trying to call a SIP softphone from an H.323 hardphone. The hardphone is connected to a Definity Prologix R12 PBX with a MedPro card and a CLAN. The Avaya is setup to send any call to extension 1609 down an H.323 trunk group that is destined for the Asterisk server. When I call

Hi all, I have problem with my Asterisk. I'm using the softphone Xten-Lite. I've removed the SIP client information in sip.conf. The softphone can't register to Asterisk, but it can make outgoing calls. I've tried to add back the SIP client information into the sip.conf, but make a wrong password in the softphone. The registration and outgoing calls are failed as expected.

Folks, I have made an inbound call. Now I want to make an out bound. I have a sip softphone hitting the pbx. On my motorola firewall I have allowed all traffic in and out to my address. There is not a firewall on the PBX yet. here is the log off the softphone..and the PBX logs look like this Registration from 'Tommy Denton <sip:201@209.101.93.30>' failed for '

Hi, I am looking for a good (not necessarily free) business-grade SIP Softphone that supports: 1) G729 2) Outlook contact integration (click on number to dial) 3) Remote provisioning (not a must, but a very nice to have) 4) Customizable skin (again, not a must but a nice to have) I've seen X-Lite (which has only 2 lines, not enough). The commercial version of X-Lite looks nice, but

Hi:I 've a question: I'm using asterisk@home; I've seen the dialparties.agi , I want to do this; I've one softphone and I want register it in 2 different Proxy;only X-lite permitted this, all others no; I want have more proxy with others softphone;I run asterisk - R and I've seen when a softphone is registered(when I run it). My question is: Who is the file who makes the

Today's "frustrated programmer" award goes to Linphone, which has the following debug output: > (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer! I get this message when I connect to linphone using a softphone, or when I try to use linphone to connect to asterisk and listen to an announcement. I suspect that

Hi all, I'm facing an issue with my asterisk server when an extension (X-Lite softphone) tries to register on it...A huge amount of packets is exchanged between endpoint and asterisk server while the X-Lite is online...Even when I sign out from X-Lite, the asterisk server continues sending packets to my machine...Can Someone help me in that? Please find the SIP packets between asterisk and

Hi, I have an Asterisk box behind a NAT address and also a Xlite 4 soft phone behind a different NAT network. Asterisk -> Nat -> Internet -> Nat -> Softphone. I can register my softphone to the asterisk box ok via SIP but the RTP stream from the asterisk box is addressed to the private non-routeable address of the softphone when I turn on rtp debuging. How can I configure the rtp

Greetings, I am new to all this VoIP stuff and have been having a bit of a hard time getting my soft phone working as a SIP client thru Asterisk. I apologize to start off with such a simple question and hope it's ok to post this and see what others have done. THE GOOD NEWS: I have successfully setup Asterisk 1.07 on an OSX machine. The build is running and working successfully. I am able

Greetings, I am new to all this VoIP stuff and have been having a bit of a hard time getting my soft phone working as a SIP client thru Asterisk. I apologize to start off with such a simple question and hope it's ok to post this and see what others have done. THE GOOD NEWS: I have successfully setup Asterisk 1.07 on an OSX machine. The build is running and working successfully. I am able

Good morning all, I'm setting up Asterisk for the first time with no prior PBX experience. I'm following Andy Powell's 'Getting Started with Asterisk' (http://www.automated.it/guidetoasterisk.htm). This is my second time through that document - as I did something weird the first time and really upset it somehow - and I wanted to ask a few general questions of the list.