similar to: Newbie Planning Help

Anyone aware of any SIP softphones that might virtualize well with Citrix presentation server? I suspect I know the answer already as I have been researching softphones that work with Cisco CallManager that can be virtualized if you will with Citrix and have come to learn that its not something that seems to be doable at this time. I have to assume that the issues affecting the virtualization of

Is anyone doing this or has anyone tried? The thin clients are running WindowsCE, a browser, and 300mhz. They are Wyse units. I wonder if anyone has any practical advise or can recommend the best phone or method to load a stable softphone on one of these boxes? Thanks, Steve Totaro

Does anyone know a softphone that can accept URLs during a call and open that page in the default browser when the call is answered? I Know DIAX and the IDEFISK, only pro version.I need another ones. It can be using the cmd SetURL Regards. -- Bruno de Assump??o Loureiro msn: loureiro_bruno@hotmail.com

All, I have a simple question and a complicated reason for asking: Is it possible to change asterisk's source port for outbound IAX2 connections? I've tried using "sourceaddress" to no avail. I can set it to: proper.ip.of.box:4569 or 0.0.0.0:4569 and it works as expected. But if I try to set it to: proper.ip.of.box:5000 or 0.0.0.0:5000 it fails around line 8536 in

Hi! Problem: I can't hear what the people at Location B i saying, they hear me but I do not hear them. They can call, I can call. Just no sound. My current setup is: Softphones/Hardphones(Location A) <-> Asterisk <-> Firewall/Nat <-> Internet <-> Firewall/Nat <-> Softphone/hardphone(Location B) I am having problems with sound, I have opened the

I've now setup SIP for: - internal softphones - registering with external providers (like FWD) for making calls - receiving calls from theese providers For the latter step, it was necessary to forward ports from my NAT to the asterisk server: 5060 + range of ports mentioned in rtp.conf. I was just wondering about how to make this setup as secure as possible. Here's what I've done so

Hi, We are using MozIAX as a softphone with a USB headset and are making outbound calls using IAX with ulaw encoding to our voip provider. We're running asterisk 1.4 Users are complaining that the ring tone generated by asterisk is much louder than the voice call once connected. They are having to turn the volume down to avoid being deafened by the ring tone, but then have an unacceptably

Some more info: Just tried this on a server without using any NAT and no port forwarding, no masquerading, and I still have the same problem. So there goes that idea. I do not know what this VNAK error means. By the way, I am using the latest version (1.2.6) of asterisk, have also tried other versions with the same problem [1.0.9 (Ubuntu Breezy) and 1.0.7 (Debian Sarge) and 1.2.1 (Ubuntu

I'm trying to set up * server behind NAT. The box is set up as DMZ in my DSL router, i.e. all incoming connections without explicit port mapping are forwarded to *. So far I'm unable to get this setup to work for either IAX or SIP (tried IAXComm & XLite softphones on public IP address). Data seems to come in fine (IAX/SIP debug shows message interaction taking place), but there is no

Hello, I'm doing some research concerning iax encryption, I haven't find any clients (softphones or hardphones) which implement so I have not tested it yet. There was also this message on asterisk-security mailing list http://archives.free.net.ph/message/20070507.101933.222987b2.en.html which got no answers and this makes me think that this iax encryption is not much interesting for the

I've tried Idefisk and Cubix Softphones, and they both work fine, except for two issues: 1. Idefisk seems to have a longer delay between the time I can hit tones, and 2. Cubix, while can send DTMF faster, never actually connects to an Asterisk-dialed call -- I can't hear the party who answers. #2 has been asked but unanswered here:

Has anyone had any luck with MozIAX (Mozphone) on FC4 with Firefox 1.6? jslib and moziax install through Firefox correctly - at least that is the message I get. I am able to log into the IAX Phone on Windows, however I get an error stating: -------------------------------------------------- FATAL ERROR: no connection to "network_client". MozPhone will stop now!

Running * with default config files except for sip.conf. Any call made is dropped 5 seconds after connection, with the following messages: Mar 17 16:37:41 WARNING[1009461760]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call 6C94C1B1-77C4-11D8-91FB- 000A95DA04DA@192.168.1.152 for seqno 48221 (Response) == Spawn extension (default, s, 5) exited non-zero on 'SIP/2000-6bd7' Mar

Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to

Thank you, I needed a starting point to start my post. 1. Asterisk 12.8.1 (IAX2 voice issues) no video issues. Voice issues on IAX2 Trunks, All extensions are SIP. The IAX2 trunks on Asterisk 12.8.1 produces only one error out of : iax2 set debug trunk on [2015-03-10 16:28:42] WARNING[9614][C-0000000b]: chan_iax2.c:1793 compress_subclass: Can't compress subclass 2097217 On the box running

I know this issue has been covered with at least 2 different patches, and probably a dozen different discussions, however I'm a bit unclear as to what my options are. I have a DSL line coming in with 8 IP addresses going to an OpenBSD firewall doing 1:1 NAT for machines behind the firewall. My asterisk box is one of these machines, and I'd like to allow foreign SIP clients

After having been toying around with asterisk and various VoIP stuff for a couple of weeks now, I want to recommend a preferred protocol and softphone to friends and family for calling me up. As SIP and H323 are such a mess to set up in NATed environments, the only reasonable protocol option right now seems to be IAX. After looking at http://www.voip-info.org/wiki-Asterisk+IAX+clients and trying

I have been looking around for examples or code on making a click to call application for web sites... has anybody had any luck on this topic? Is there any open source code out ther that could do this? Regards AK

Hello there, I'm a new Asterisk user and I'm having difficulties to connect to and from my Asterisk server. Can anybody give me a hand? Here's some background information: * I'm running RedHat Linux Enterprise 4.0 * When iptables is stopped, my server can register with IAX service providers and receive registrations from IAX softphones. However, it does not succeed in

i read that asterisk supports iax,sip and h323 protocols.... i've used sip & iax softphones ... now i've a hardphone... an IP phone (Netphone) that supports h323 ..... i've compiled pwlib ,oh323 and asterisk -oh323 successfully ... but i m unable to place calls to/by my phone... i m confused whether to use h323.conf or oh323.conf and how ? i think it's different from iax.conf