similar to: Using Asterisk with FWD through NAT

Same thing. It will make sense to try Register => <FWDnum>@fwd.pulver.com:<FWDpass>@fwdnat.pulver.com:5082 but in that case Asterisk sends REGISTER sip:fwdnat.pulver.com SIP/2.0 which is not right. It should be sip:fwd.pulver.com but sent thru fwdnat.pulver.com:5082 BR Borut -----Original Message----- Subject: Re: [Asterisk-Users] Using Asterisk with FWD through NAT From:

I am having trouble with this. I can dial 1800 numbers fine as well as FWD service numbers but not Vonage. I can be called from ipkall and fwd and can call aixtel numbers. I use aix2 with Fwd. My extensions.conf for Vonage: ; vonage numbers ; ; +2431 exten => _2431XXXXXXXXXX,1,SetCallerID,${FWDCIDNAME} exten =>

Thanks for taking a look greg and hank. This seems to be getting bettre everyday..help please My sjphone is running on the same box as asterisk...i believe then the red hat firewall should not be a problem. Whenever i dial from CLI i get ######### Executing Goto("OSS/dsp", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("OSS/dsp",

Hi all, I'm unable to put app_festival to work. I successfully patched, installed and tested festival (interactive logon and telnet to server port) which seems to work without problems. But when I test it in asterisk I got the following trace in console: -- Executing Answer("SIP/bsenicar-850b", "") in new stack -- Executing

Hi , Thanks greg , for pointing out the valuable resources for reference. I tried SJphone in a windows environment to connect to fwd and it worked fine(including (audio). Now have to do the same thing for linux(red hat 9 ) and hope the nat issue is resolved. Now i would like to connect asterisk to fwd and instead of the SJ phone connecting to fwd directly i would wish to connect through

I'll admit it. I'm a asterisk newbie (but no stranger to telephony). The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone specials) on a private segment calling to a Linux box acting as the segment's firewall with a leg on our public network. The phones are setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks to the Asterisk HOWTO). Getting IAX

I can't get my Asterisk to register/place calls with FWD. Here's what I have in my SIP.CONF: register => 11111@fwd.pulver.com/11111 [fwd] type=friend secret=somesecret host=fwd.pulver.com username=11111 fromuser=11111 fromdomain=fwd.pulver.com I'm using CVS version of Asterisk, checked it out last week. I get authenticate error when registering with fwd, and all my calls to

Hi! I've set up a Samba server (running winbind) as a Win domain member server. Users authenticated from win domain and mapped to NIS uids and gids. Everything was working well, but the next morning winbind stopped to respond. I reset it, so now I get wbinfo, ypbind (ypcat) results ok as before, but users suddenly cannot authenticate any more. My <client>.log files are empty,

Someone said something about FWD registrations requiring the extended syntax shortly, like this: register => 12345@fwd.pulver.com:somepassword@fwd.pulver.com/12345 At the moment, using the extended syntax doesn't work - I get authorization errors. So stick with the old syntax: register => 12345:somepassword@fwd.pulver.com/12345 JT

I'm having a problem trying to write a simple vorbis file player using DirectSound. The decoding portion is pretty much straight from the vorbisfile sample code. The pcm data gets put into one half of a DirectSound buffer, and as that half is playing, the next half gets filled. The problem is, there's skipping and some noise when the file is being played (although some of the music is

[ Sorry, I incorrectly copied some Reference headers into this post and tacked it onto the wrong thread. -Steve ] So far, I have been able to receive incoming iaxtel calls via my assigned 1-700-xxx-xxxx number, but only when using md5 authentication in iax.conf: [iaxtel] type=user ; Incoming calls only context=incoming auth=md5 secret=<mysecret> ; Required for

Sounds like either your machine is too slow or you have some other buffer issue. A decode thread is usually ideal so that data can be decoded before it is needed by the buffer fill operation. You might be able to get a sasifactory solution by increasing the number of buffers and using them in a round-robin fashion 1-2-3-4-1-2-3-4. Wouldn't do any more than 1/4 sec per buffer. Cheers, Chris

We have an Asterisk server acting as a hosted PBX system for many clients, and we're going through an upgrade to Asterisk 1.6 by moving our most important (and complicated) clients one at a time. But we're having a problem with one customer that I really can't explain. I can place calls directly to one phone at the customer's location (they also have an IVR that asks for an

Hello, Does the "lda" function (package MASS) perform or can it perform classic two-group Fisher discriminant analysis? R-version: 1.9.1, MASS package (latest available) Thank you, Borut Rajer

Hi, I've posted a simular message little over a week ago so sorry for reposting. I need to register to freeworld dial up from behind a nat. Using the xten software sip client works fine but with asterisk I don't know how to do it. Last time I posted I got different responses. Some saying I can't register with an outbound proxy from asterisk others said they have done it. If it is

OK, finally I have made cbq run ! now I would like to know if it is posible to limit a range of IP to a speed. something like this : from 192.168.1.1 to 192.168.1.21 limit to 128K I don''t whant to limit each IP to 128K, what I would like to do is limit all 20 IP to have a max of 128K. Is it posible?

Hello! I've got users that access linux file server from Win stations using Samba and winbind authentication but also have AIX based workstations accessing the same server using nis authentication. They have the same username registered in NIS domain and in Windows domain. Is it possible to map winbind users (usually using uid & gid > 10000) to nis users, so user john

Hi, Is it possible to use asterisk with Free World Dialup (FWD) ? Did someone manage to make it work? how? Best, -P -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup CareerBuilder.com has over 400,000 jobs. Be smarter about your job search http://corp.mail.com/careers

Hi there, It might that this question was already answered but I can't really find any proper documentation for this. I am in a process of migrating about 70T of files from our Win2016 storage server to a Samba (2:4.17.9+dfsg-0+deb12u3) server. Whilst doing this I want to preserve all the ACL's which were set on Windows shares. Is there any related documentation around this topic how to

I am trying to use festival (latest version 1.4.3) I have downloaded all the files needed and patched it with the provided diff. festival does work and does tts fine. but when I call Festival either from an extention or an AGI script, I get this in my asterisk messages log, but no sound on the channels (H323 or SIP) - they (the clients) just say "trying" and then hangup... Sep 24