similar to: is asterisk a good solution?

On 02/15/2018 03:44 PM, Joshua Colp wrote: > On Thu, Feb 15, 2018, at 6:43 PM, thelma at sys-concept.com wrote: >> I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports >> >> IN audocodes setting I have: >> "EndPoint Phone Number" >> >> Channel: 3 phone number: pstn-4444 >> Channel: 4 phone number: pstn-9998

hello list , my set up is like this ip device -->ser ---> asterisk(astcc) --> pstn gatewsy my asterisk version is 1.0.2 iam using the ser as registration and asterisk aa the prepaid one with the help of the astcc. now my problem is the destination people i.e the pstn line s are listening low voice and also the blurr sound quality along with the audio of the ip device at

i installed yesterday asterisk cvs version and tried to register asterisk as a sip ua with my proxy. there was two problems that i was not able to solve with the style of register line shown in the default sip.conf: register => foo:password@proxy.com the first problem was that asterisk converted the host part of the to/from uri from the domain name proxy.com to its ip address, which i

On Thu, March 26, 2015 22:29, Michelle Dupuis wrote: > You have to consider whether you really want "anonymous" calls, or you > just want to enable SIP calls from trusted companies/partners. The > latter means setting up routes to these companies and (ideally) > registration between peers. > This is what I am trying to get a handle on. It seemed to me that the promise

I'm using Audio-codes MP-114 unit and it has two public lines PSTN ports IN audocodes setting I have: "EndPoint Phone Number" Channel: 3 phone number: pstn-4444 Channel: 4 phone number: pstn-9998 When I am calling " pstn-4444" the port number "Channel:3" lights up but asterisk is showing that the call is coming on "pstn-9998" -- Executing .....

Do some reading about contexts in *. Basically, you want all "public" sip requests to land in a dialplan context that has no access to PSTN, and requests from your own SER box(es) to land in another context (that DOES have access to PSTN). You can achieve this by adding an entry to your sip.conf for your SER box with it's IP address (and context) specified. ----- Original

We have a FreePBX-12 / Asterisk-12 setup that supports about 24 extensions, most internal Snom870s but six or so external (Jitsi-2.8). we use TLS and SRTP everywhere on our side of the fence. The server host is a dedicated atom(tm) box using the FreePBX distro (CentOS-6.x) and is up-to-date. Registrations require very long random passwords and registrable devices are further restricted by

I've already sunk several hours into this without any real progress, so I'd really appreciate any help My task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569

What are the alternatives to FWD with IAX2 registration capability. FWD is great, but their IAX2 is not the priority and if it goes down it takes days to restore it. I want to use IAX2 protocol but the end point (Sipura unit) need to be able to register over SIP behind firewall. Line1 is registered with FWD PSTN need to be registered with somebody else. What are my alternatives? -- #Joseph

Hi My call flow is T38 static IP gateway --> Asterisk --> Sip Provider--> PSTN Call is placed in reverse direction - from PSTN to T38 Gateway. T38 GW reinvites with T38, and asterisk passes it along to the SIP provider. The SIP provider challenges it and asterisk reponds to the Challenge with INVITE with Auth credentials...however, the Asterisk changes the SDP and replaces the T38

Hello list, When you have an asterisk box connected between the VoIP phones and an PSTN gateway what is the role of asterisk. Proxy server: stateful or stateless? From what i read in the: "Understanding the SIP, second edition" from Alan B. Johnston i think that asterisk is a stateful proxy server as well as registration server. Am I right? Can asterisk be configured to work as

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031207/e5e9b8eb/attachment.htm -------------- next part -------------- Hello all, I am newbie to Telephony world (IP and PSTN). Please excuse me if you find my questions very dumb. I am trying to configure my IP 7940 with the Asterisk, when phone boots up it only shows the message "Phone

Hi, Sorry for my bad english but I?ll try to explain my problem I got an Asterisk running in my house with ADSL... I?m using X100P and TDM400P cards.... My intention is get calls via PSTN to my house and Redirect to my computer in my work using X-Lite by SIP... Here?s the map with Firewalls Call for anyone to my house => PSTN => X100P => EXTENSIONS => SIP/RTP => ISA MICROSOFT

Just been contacted by a UK Enum registrar looking for ITSPs to become resellers of their Enum registration systems ... Is anyone using Enum? Does anyone (other than cynical old me) think that Enum is a spammers best friend? Has anyone received a spam VoIP call yet? (ie. one placed directly over the Internet aimed at a SIP URI to a PBX which allows anonymous incoming calls?) I can see that

Arrggghh. Tearing my hair out here. I'm trying to set up the spa3000 in the UK for my home, and want * to control the dial plan I've googled to no avail. I've read the manual to no avail. Can someone, please let me know what the parameters is the spa and * are to a) receive a call from the pstn b) make a call to the pstn from the phone attached I can make sip to sip calls (i.e. I

Hi, as stated in the documentation, it's allowed to set FAXOPT(faxdetect)=yes/no to allow fax detection. It's done (see below) but still fax detection :-( Extension 300 is hylafax with iaxmodem. On the upper Asterisk gw it's the same, despite the faxdetect set to no we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile phone calling the 0123456789 PSTN number.

FYI It looks like FWD is looking for value added service ideas for free as a volunteer. I think it will fail but we shall see. I really don't get the nerve of them (Free World Dialup has changed it's name to FWD) to ask for free ideas and development on a non-free service. Maybe if they can come up with a killer app and people will adopt it, then it might work, but then again, people

Hi all, I'm planning to build a VOIP solution for handling SIP calls coming from endpoints registered on a specific SIP proxy...I made some research regarding network architecture and found out that the best solution is to use OpenSips as SIP proxy for registration and local calls between registered endpoints and use asterisk server with a2billing for PSTN calls, rating, routing and all other

Hi All, Downloaded latest version of Asterisk from www.asteriskwin32.com and installed on Windows 7. Here is my sip.conf [general]context = demo ; Default context for incoming callsbindport = 5060 ; UDP Port to bind to (SIP standard port is 5060)bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)srvlookup = yes ; Enable DNS SRV

Hi all, So I have been reading through the docs available online and the different threads on this list, but I cannot seem to get this phone to work. I have configured the OS79XX.TXT and SIP/SEP*.cnf files (see attached), when I configure the phone to point to my tftp server and reboot it I get this message: Connection received from 10.6.0.224 on port 50608 [09/02 12:16:11.750] Read request