similar to: outbound proxy field in sip.conf

I know asterisk isn't a real SIP proxy and is more of a multi-protocol pbx with limited SIP support, but... ... is it possible if you have a central registration server that handles all of your dialplan routing and several asterisk PSTN gateways that it routes calls to for an outbound SIP conversation using reinvites and NOT have the registrar box try and send ANY RTP traffic back to the

Hi, We're deploying a small VoIP solution for a group of teleworkers. Naturally, this exposes us to all sorts of fun, most of which we seem to have working properly. However, some NAT issues are still bugging us and we have noticed that often these situations didn't exist when users were connected directly to our VoIP provider, voiptalk.org. They have something which they call a

Hello, I have had numerous issues with PJSIP outbound calling in Asterisk 13.1.0 and SIP.US SIP trunk. My Asterisk server is on EC2 and I have opened up the appropriate ports. The SIP clients can be anywhere on the Internet, including behind NATs. I am able to get to my Asterisk server's internal extensions via the DID (and appropriate dialplans) but I am not able to make outbound calls to

I'm using the latest SIP firmware from Avaya. The phone receives the 46xxsettings.txt OK, and then after entering extension and password it goes to the home screen saying 'Registering'. When I check options->ViewIPSettings->IPAddresses on the phone, the registrar and SIP Proxy fields are blank. I have both lines: SET SIPREGISTRAR "67.1XX.XX.XX" and SET

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

Hi folks, I want to fix hole in my asterisk set up. I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN, Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go 'other' places. This senario works fine. Now the issue is someone else running a vocal or another SIP proxy can redirect his calls to my * as well. Those calls two will come through general

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 When connecting IAX (gnophone) to SIP (kphone) or other way Asterisk acts as a proxy, but when connecting SIP to SIP it works only as 'SIP registrar' forwarding SIP requests to client. Is it possible to make Asterisk work as a 'proxy' so that any incoming calls would be ade to Asterisk and then internally forwarded to receiver?

Hi, I have been experimenting with NAT and Asterisk a bit now. Though I have made progress along the way, I have come across the following problem. I'll really appreciate if anyone can provide me any help or pointers. Thanks! Successful Scenario: ------------------- All sorts of NAT calls are successful with full two-way media when both end points are locally subscribed users. Problem

Hi all, I encounter an annoying problem using Asterisk. I 'm using SIP. I try to register an Asterisk as a SIP end user with another Asterisk. If I put both asterisk in the same local network, no problem to do it. The asterisk end user registered perfectly with the other (let's call it the registrar). Authentication is enable and works fine. The problem occurs when I put the registrar

I'm getting the below error when trying to compile the latest git. gcc -c -I. -I. -I../../include -I../../include -D__WINESRC__ -D_REENTRANT -fPIC -Wall -pipe -fno-strict-aliasing -Wdeclaration-after-statement -Wstrict-prototypes -Wtype-limits -Wwrite-strings -Wpointer-arith -g -O2 -o registrar.o registrar.c registrar.c: In function 'DllGetClassObject': registrar.c:747: error:

Hi, I've been building wine from source for years, originally because I needed some custom patches, nowadays out of habit. Just now I'm trying to build 1.3.8, but I get an error during build: make[1]: Entering directory `/home/peter/src/wine/dlls/atl' gcc -m32 -c -I. -I. -I../../include -I../../include -D__WINESRC__ -D_REENTRANT -fPIC -Wall -pipe -fno-strict-aliasing

Hi, all, I am a beginner of asterisk SIP, now I have 3 pc, one runs asterisk as a SIP proxy, and the other two run softphone(Ubiquity) as User Agents. As below: User Agent <------------> Proxy Server <----------------> User Agent (Ubiquity) Asterisk SIP (Ubiquity) My sip.conf and extensions.conf is as follows: sip.conf [general] port =

Dears; I do not know if any had experience in using speex or ilbc with IAX and got good results, because I am facing a problem with GSM. I am facing a noise problem when I am using GSM with IAX trunk as following: IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk using GSM codec ---> Remote Asterisk Box ---> Digium Card (FXO) to terminate the call to the destination While no

Hello List I work at an SIP Provider and we have added and SBC in front of our Voice Switch to protect it. This requires all our SIP Trunk customers to register via a 'proxy'. I struggle with Asterisk to work over a proxy. This is what I have done so far. register => username at sip.example.com:password at sbc.example.com This works fine, asterisk is sending registrations via the

Forgot on the Snom 200 it won't reboot if under the Memory tab in the web interface, Connections > 0 then remote reboot is not possible. Manually cycling the power allows the phone to be rebooted by Sipsak remotely. HOWTO: Reboot a Snom with Sipsak Checklist: 1. Under Advanced in the web interface, is Network Identity set to 5060? 2. Under Advanced in the web interface, is Challenge for

Hi, I've got a (maybe stupid) question: I'd like to install Asterisk on my notebook (just to play around with it). Is my internal phone modem equivalent to an FXO port? Hence, could I make phonecalls to my pots-line with it? thanks philipp

Hi, I've got a (maybe stupid) question: I'd like to install Asterisk on my notebook (just to play around with it). Is my internal phone modem equivalent to an FXO port? Hence, could I make phonecalls to my pots-line with it? thanks philipp

Hello, My colleague installed a Asterisk home as company's SIP server and I would like to integrate the Quintum gateway (SIP) but the calls don't get through. Bellow is are the configurations on each side: Quintum ******** Primary Registrar = 202.69.190.244:5060 Primary Registrar User Name= sipquintum Primary Registrar Pwd= sipquintum Primary Proxy =

On 5/4/17 3:43 PM, Michael D. Setzer II via Syslinux wrote: > Unable to determine IP address from host name "www.hdt-project.org" > > Getting this today? Not sure what issue is? > I paid for the renewal back in 08/04/2016 and and in 2015, so the domain > should be current? But the whois seems to show it is expired? Went to the > gandi site, and it doesn't show a

Unable to determine IP address from host name "www.hdt-project.org" Getting this today? Not sure what issue is? I paid for the renewal back in 08/04/2016 and and in 2015, so the domain should be current? But the whois seems to show it is expired? Went to the gandi site, and it doesn't show a renewal option or anything? whois hdt-project.org [Querying