search for: freestylenetworks

Hello all, I updated from CVS 3 days ago and now my IAX2 gateway is dropping calls without warning. It happens right in the middle of a conversation with no pattern. I never had this Problem before and am usually talking 2-3 hours a day. Is their a bug? Should I rollback? Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul

Is it possible to allow distinctive rings work for FXS ports as well? I need a certain FXS extension to ring a distinctive double ring. I modified zapata.conf appropriately for dring1,dring2 and it just Seems to ignore my updates. Do distinctive rings only work for FXO ports? Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul Seniuk.vcf

I was looking at the wiki on 'Asterisk as a voice/fax switch' And was wondering if the extension 'fax' is global to extensions.conf Or just to the context it is in? The reason I ask, is that my PRI might have 5 channels that will be scrictly Fax, and to be functional, I need multiple 'fax' extensions in my various Contexts. Hope that makes sense, Paul Seniuk

Has anyone has issues with echo using a Wildcard with a PRI from a major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom). We are using a T1 from GT that is giving use annoying echos whenever a SIP/IAX2 client calls a local analog line. Calling cells phones is no issue since its digital. Regardless, there should be no issue with echo on a PRI at all. NOC at GT is telling us

I am attempting installation of spandsp on to an Asterisk installation on Linux RH9 the distribution i am using is that are URL http://ftp2.tootai.net - the README for which i have followed verbatim - my only issue on this was the target for the port.h / tif_dir.h / tiffiop.h files in the 'headers' folder of the distribtion i put these in the /usr/include folder based simply on the

I should also mention that Accounting rules are not stateful -- each rule only handles traffic in one direction. So for example, if eth0 is your internet interface and you have a web server in your DMZ connected to eth1 then to measure HTTP traffic in both directions requires two rules: DONE eth0 eth1 tcp 80 DONE eth1 eth0 tcp - 80 Associating a counter with a chain allows for aggregation.

Helo to all, I am attempting to establish an IPSEC tunnel to a remote freeswan G/W with my laptop. My laptop sits in behind shorewall at home. From the documentation, this is what I Modified in Shorewall: /etc/shorewall/tunnels: ipsec loc 24.65.x.x /etc/shorewall/policy vpn loc ACCEPT loc vpn ACCEPT My question is, have I left anything out?

Hello, Can Asterisk perform as a H323 Gatekeeper? Here is my scenario: I have a customer that has a calling card program that will be transmitted as VOIP from a Cisco 5300 in Hong Kong and terminated here in North America. The catch is that, the termination is being handled by a third party company and we still need be the 'buffer' in which we pass the VOIP traffic to the

Hello all, I am trying to setup a local asterisk server with SIP & ZAP extentions with And IAX2 'switch' to another Asterisk gateway with a PRI. I have managed to Get it working correctly for calls coming in and out. However, CallerID only Seems to work when I dial out. However, everytime I receive a call, it shows Up as 'IAX Guest User' on a SIP client. How can I

Hello all, I have a WildCard TE410P setup and working with a full PRI with the latest CVS. SIP and IAX2 gateways are accessing the PRI without issue, however; echo is very prominent in some calls and is only heard by the IAX2/SIP client. The echo Is not present in calls to cell phones because they are digital, centrex land Lines have a barely noticable echo, but analog lines aint so pretty.

Hello all, I have a TE410P hooked up to a single PRI. Incomming CallerID is fine, and Outgoing works as well. However, if I change my dialplan for an extension To do a 'follow you, follow me' or setup an auto attendant that rings extensions Thru to cell phones, the CallerID always shows up on the call reciever as '708'. When I do a verbose dump at the console, it appears that

Hello all, I updated from CVS a few days ago and noticed that my calls just cut out without reason. The CLI says this: -- Hungup 'Zap/3-1' It occurs without error or warning. Is their a bug in CVS asterisk or libpri? This never occurred before. Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul Seniuk.vcf Type: