similar to: multiple sip phones behind firewall

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

Ok I have a question. Seen it come and go around the mailling list for a while but never really seen an answer that seems to sort it out. What is needed is some interface from * > Mobile Phone > Mobile Network Service. At this point all the providers in AUS that I have found are charging a Premium Rate for Land Line > Mobile Network services. What I would like to do is be able to

Hi all I installed *wine 9.6 on fedora4, IE6* , and *APPs* for *CALC TI83 plus * i tried to use the *USB *cable to connect to the *CALC* but did`t work then i used *serial cable* after i make *? ln -s /dev/ttyS0 ~/dosdevices/com1? *then i run the "TI Connect APPs" to coonect to the *CALC *it finds the *COM1* when it tried connect to the* CALC* but connection filed. any one

Is it just me or are there problems? The list has just shutdown over the last 24 hours :( David

HI ALL; Is there anybody who use app_radius(astersik radius module)??????????? is it stable? Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040803/8a096bfe/attachment.htm

Ok I have a challange that I can't seem to find a way to fix it. My Voicemail in * timesout after 30secs without fail everytime no matter what I do. I have incomming calls comming in through Freshtel IAX2, if it goes to SIP extension when it is online it can hang on for what ever time the call goes for. If however it goes to the Voicemail it will timeout at 30sec and I can't seem to

Does anyone know how long the orders take? I ordered some a couple of days ago and it said normally 24hours, and I am guessing that the weekend cause's some delays but it did not say anything abouy that. Any one got any ideas on how long generally over the weekend it takes? Thanks David

Just a couple of questions for Aussie users/resellers! I have only just started to look at asterisk a couple of weeks ago and have found some very intresting discussions and some useful info on what can and can?t be done with it and technology. The questions i have is, Are there people using it is Aussie?( I would say yes so prob answered my own question ) :) Hardware, is the digium

Hello, Have a weird problem when using asterisk (1.0.6). There are certain numbers I cannot dial when using asterisk with my broadvoice account. No problems with inbound. With outbound calls, I can call some numbers (for example broadvoice customer support number) and unsuccessfully with some. However, when I configure my account directly on x-lite, I dont see these outbound problems. Here is a

Hello Sorry for asking this question..im new to the list. Where can i get the latest samba for linux...i currently use 1.19.18p5. What are the added features? How can i make my samba server seen at a different domain? My server is in the 10.36.x.x domain. It cant be seen elsewhere i.e. , 10.3x.x.x or 10.x.x.x domains. I am not subscribed to this list. Hope someone helps Cheers Paul

Now I have searched around and not seen anything to do this. I want to in remote locations were we need to have single or 2 PSTN lines for in dial as little hardware as possible and as stable as possible so that they will operate without user intervention. What I want to do is be able to take a single PSTN line in and go out through adsl for the Inet link. These would be in VERY remote

I have been tring to send email to the List for the last day or so and have not seen it come back on the list and have not seen any reponse's to my email so I am unsure if it is making it to the list. And I have also been seeing some of the same emails over and over again. The list does not seem to have got back to 100% since it was down :( My mail server works fine I am not having those

I have seen the list of codecs for the ATA 186's but not sure if it was 100% or not. I want to know really is it possible to run GSM or ilbc on them or is a G729 lic the only way to get a low bandwidth codec? This is the list of codecs that I have seen. RxCodec and TxCodec?Configure the codec ID. * G.723.1?Codec ID 0 * G.711a?Codec ID 1 * G.711u?codec ID 2 * G.729a?codec

Ok I upgraded tonight a server from CVS in Late NOV to one just downloaded tonight. It all runs up OK and I can contact it from my ATA 186 using g729a codec and that all works fine. What I am having trouble with is connecting through IAX ATP.org.au in AUS to my server. The connection comes through OK I can see all the tracking info in the console OK but I get 0 audio in either direction.

Hello Guys. I recently contacted polycoms tech support asking if their phones supported XML pushed information to which they replied that only model 600 had a microbrwoser capable of reading dhtml files and such. My question to the community is: is somebody doing any XML info push to any brand of phones except Cisco? How are you doing it? One of the wonders of VoIP should be the means to send

I am playing with the dialplan to get it working and I have a challange with this error. I can't find what it means on the wiki :( Any sugestions would be helpful at being able to forward it to the SIP phone if it is online and avaliable but then let that fail and drop into voicemail if it is not online or is busy. cheers David -- Executing Dial("IAX2/firefly@89280250/3",

Dear All, I have the following CPU info on my asterisk server: Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 EST 2008 i686 i686 i386 GNU/Linux I need to install G729 on the asterisk server just to pass through and not for encoding...Which G729 package do you advice me to install? I tried several packages with no luck Regards -------------- next part -------------- An

Greetings list, I'm coming across an issue with some of the GXP-2000 phones we have out in the wild at clients' employees' homes. In most cases they're behind consumer ADSL NAT routers on a dynamic IP from their ISP. In a nutshell, the phone is unable to be called unless it's restarted first, after which it's fine for a good few hours, then it stops working until

Hi, I'm trying to configure Sipura 2000 (behind NAT) which connects to Asterisk (public IP, no NAT) and having interesting results. When Sipura is behind Linux/NAT firewall it works great and no special NAT settings on Sipura are necessary. The issue I'm having is when Sipura is behind Linksys broadband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear

Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server -- #Joseph