similar to: SIP Registration with Entice Softswitch

Hi, I'm considering submitting a project for this year's Google Summer of Code program that would involve making progress on hardware accelerated video decoding for the nouveau driver and wanted some feedback. I know that this isn't a big priority at the moment amongst the regular contributors, but I think it would make a good GSOC project because of it's somewhat narrow scope. I

Hello, I have 1 SAMBA PDC : version : Samba version 2.2.6pre2 name: lothar ip: 192.191.193.10 and 192.168.1.2 PDC of ADMIN01 I have 1 SAMBA client: version:Samba version 2.2.7a name: intranet ip 192.191.193.2 member of ADMIN01 with option Wins server and winbind . and one Win2000: member of ADMIN01 --------- My pdc have the add user script (work fine i have see the machine in /etc/passwd and

I am v v newbie, want to get to grips with linux but mostly interested in using the PC; so the idea of wine appeals a lot. from what I see you generally have to build or compile or something it. Is there a rpm package, if I am using the term correctly, like a windows self-installing thing, i.e. one single file I can download, click on, it goes thru a (preferably non-interactive, I am happy

>>>>> Avraham Adler >>>>> on Mon, 13 Jan 2020 14:38:12 -0500 writes: > Those of us stuck on Windows but who attempt to develop properly are > wounded to the quick, sir! > :) > Avi Indeed, you had a ' :) ' , but others have perceived this as an insult. I'm really really sorry for that and do want to apologize to all of

Hello All, I've been trying for some time to get Asterisk to register with a remote SIP gateway. I?ve recently managed to configure an SJ Phone to work with W2000 so know the configuration parameters work correctly. Asterisk doesn't authenticate properly and I notice that the authentication request appears different to SJPhone's. Do any tools exist to enable me to check these

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Welcome to the first release of the 3.1.0 development branch. This is an unstable release series intended only for use by testers and developers. We make no claims as to whether it even works. The purpose of the 3.1.x releases is similar to what we've used the 'alpha' tag for in the past. These are unstable, development releases that

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Welcome to the first release of the 3.1.0 development branch. This is an unstable release series intended only for use by testers and developers. We make no claims as to whether it even works. The purpose of the 3.1.x releases is similar to what we've used the 'alpha' tag for in the past. These are unstable, development releases that

Summary: We need to find a reviewer for our implementation of Ball-Laurus path profiling. It is well known that path profiling generates more precise information about a program's behaviour than edge profiling. We are conducting a research project with the goal of developing a methodology to make feedback-directed optimization (FDO) more sound. We are developing combined profiles that enable

Hi All, I've been experimenting during this weekend with asterisk as a softswitch, talk about me being completely lifeless, but let not talk about that. I've been conducting some really funny tests, trying to get an optimal SoftSwitch functionality. Here is my current setup: Source: Windows XP Pro + SJphone Box 1: Asterisk running in PassThorugh mode Box 2: Asterisk running in

Hi, did anybody ever try to make asterisk 'switch' calls on pri interfaces? My goal is to set up * to 1. wait on incoming calls until an extension is matched, 2. dial out and then 3. "pass through" the signaling until the call is either established or released I found that asterisk would first pick up the incoming call and then dials out. For my application however this

Has anyone setup SIP services with ss7 and lis trunks? If so .. what was used hardware and software.. we're trying to do a SIP -> pstn setup and not having much luck as QWEST keeps pushing dates off (aka trying to screw us over) for our pri lines due to the recent court and fcc activity in regards to unbundled switching and I'm looking for solutions/ideas involving SS7..

We would like to know if you have any recommendations for softswitchs to be used by a small size telco with sip services in *.

Im trying to get my asterisk box to register to a sip provider without much success. here is my console output in asterisk Dec 13 12:57:17 NOTICE[213005]: chan_sip.c:3982 sip_reg_timeout: Registration for 'voicemail.nexband.com@metaswitch.nexband.com' timed out, trying again -- Got SIP response 403 "From: URI not recognized" back from 208.149.73.5 Urgent handler in my

Hi All, I want to use Asterisk for VoiceMail for a softswitch. I can dial in to leave voicemail and retrieve. Now there are many SIP Endpoints registered to the Softswitch. The Asterisk is sending a NOTIFY msg to the Softswitch on <ip addr>:0 Somehow Asterisk Looses the port from where the INVITE came in, this NOTIFY msg is not going out of the Asterisk, I cannot see in Ethereal.

As an example....if we have a call that: 1. originates via PSTN line to one of our local DID's in Seattle 2. comes into our Asterisk server in Los Angeles or Denver 3. is routed by Asterisk for termination back to a different Seattle PSTN ....and if our VOIP call termination provider requires (in order to get their best rate) all calls to go through their Nextone

Dear all what is softswitch what is difference between asterisk and softswitch ?? regards satish patel --------------------------------- Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All; Alot of softswitches or PBX's does not accept to manipulate any SIP call without being registered firstly. So that means, I need asterisk to register firstly then I can route my calls to that SIP trunk. In IAX2, we use the register => , so what shall we do in Asterisk? And how its format will be (if we will use register)? Or what is the solution? Regards Bilal

Hello averybody, I'm looking the softswitch in digium website, anyone test the softswitch? Best Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071202/2440f782/attachment.htm

Hi, has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto or more info about needed Asterisk SW and setup ? Thanks in advance, regards, Rob.

Dear All, First, I am not so much experienced in Asterisk. I need asterisk to route the call to soft switch when the caller is not in its extensions list. And also when routing to soft switch, a number 4327 has to be added in the caller's number and then routed. I think its not so hard in asterisk. Please help me. Regards, Pratik -------------- next part -------------- An HTML attachment