search for: ix66

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Hi, Has anyone got incoming IAX to work on the above router. I can call out, but incoming calls are not reaching the * box. Has anyone got this working? Could they give me some configuration hints. Thanks John

Hi, I've just (earlier today) updated from CVS so that I can apply the dtmf caller id patches. Unfortunately this has had an undesired effect. I have an intertex ix66 which up until the CVS update allowed me to register my * server with the ix66 for my local domain (eg sip.mydomain.com). Now it appears that asterisk gets totally confused and tries to register with itself! Anyone got any ideas? Thanks Andy 11 headers, 0 lines Reliably Transmitting: REGISTER...

...ess. My two end points are an Xten softphone and a Mitel 5055 SIP phone. Both peers seem to register with * but I cannot call to one another. When I dial the associated extension, the call goes to the programmed voicemail extension (busy) yet if I create an extension to call out through the proxy (IX66), I can still reach my destination. It's just calling within * there is a problem. I suspect it's because the status is unreachable but I'm not sure how to fix it. Here is the sip show peers output. Name/username Host Mask Port Status sipset/sipset...

...for 6 phones.. I guess I would have to allocate a seperate port to each.. Also this could have major security implications.. I could use a STUN server but I have not found a free one yet.. I could use an ADSL router with a built in SIP proxy/registrar but the only one I have found is the Intertex IX66 and they seem quite expensive.. Anyone got any ideas or thoughts?? or even better experiences.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

i cannot see caller ID of the call originated from outside zap channel. i hv configured both zapata.conf and extensions.conf. i m right now in india i think asterisk only supports Bellcore enable caller ID. so is it the same bug of BT caller ID problem in UK? or it is the bug of my asterisk configuration? i hv enabled callerID from my TELCO. -------------- next part -------------- An HTML

Hi all I have 2 snom 100's and an ix66 (sip aware firewall) set up with asterisk. I needed to register a number of lines so what I've done is make asterisk register all the lines i need (attaching them to an extention eg 1000) and then register each phone with asterisk. so for example in sip.conf: register => andy@sip.mydomain...

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

...the pic that I email the guy and he send me the PDF with all the details you can find it here http://mike.calle69.net/IPC5000-brochure.PDF, and its almost the same price as Wisip. But when I ask if this phone will work with asterisk I got this answer "We didn't tested on Asteriskt but on IX66, hotsip, Cisoc etc... However, The IPC5000 should work on other SIP platform without any problem as it is standard based." I just dont want to spend 290 USD for a phone that wont work and that no one seems to use here. So I would like to know if anyone of you guys had try out this model or se...

Hi! I'm trying to set up the possibility for users to call my Asterisk from the net. The Asterisk is behind a Intertex IX66 in which a have set "Static domains" so it forward all calls for my hostname and external IP to the * box. when somenone calls at lars@<external-ip> it all works, but if they call lars@<hostnmame> it wont work. At my Asterisk console I get the following output; --<SNIP&gt...

...ot permitted. I get it on calls to the PSTN through my X100P (clone) as well as call connected through my IP telephony provider. I have also tried SJ-Phone and it happens with that as well. At the moment my asterisk server is on a public IP adress, and my client connects to it through an Intertex IX66 router. Before getting the router, I had dual NICs in the linux box and connected the client directly with the same problems... I've searched the lists but haven't been able to find any good answer, so any help is greatly appreciated :) /Anders -------------- next part -------------- An...

Hi all, I was wondering whether any of you have experience/info on Cable and/or ADSL modems that would come together with a SIP phone adaptor. What I am interested in is something that would plug directly into you ISP's cable (be it ethernet or adsl/phoneline), would combine a modem/router/nat such that on the other you could simply plug in your RJ-45 cable for your PC and a RJ-11 cable for

...LAN and the outside world. An outbound SIP proxy would take care of all SIP messages between your LAN and the outside world, the Internet. This proxy could also handle conversion of IP addresses, and assist the firewall in opening and closing RTP/RTCP ports to let the call go through. The Internet IX66 works like this, and there are ways to configure the SIP Express Router to be part of the Firewall DMZ, handling the SIP messaging and working with an RTP proxy to handle the audio. Currently, there's no support in Asterisk for an outbound SIP proxy. I am working on it, but need some help. Che...