similar to: looping back calls

Hi Everyone: I have a THREE questions. What is a sip proxy and what is the benefit of having one with Asterisk? I am well aware that we have a sip channel in Asterisk and that we have SIP registration. I am not sure why you would need a SIP server and OR a registration server. Second question, with Asterisk are you able to do video on VOIP video phones? Last question, does

Hi all, I am new to this list... Can I use Asterisk as a SIP Proxy and as a SIP to H.323 translator? I want to implement PC-to-Phone calls in the following topology (for signalling): SIP Softphone --> Asterisk --> Gatekeeper H.323 ---> Gateway H.323 ---> PSTN The RTP audio packets would go direct through Softphone to gateway. Does someone have a configuration file example of

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

Hello All; After trying A2Billing and certainly when the trunk is analogue lines (FXO ports), I faced a problem that the channels were not hanged up properly from time to time which cause us to do restart for the dahdi. Without A2Billing, I was able to handle the Dial scenario properly and no hanging for the analogue channels and no need to restart dahdi from time to time.? Really I would if

Hello; I am facing a trouble with A2Billing when using analogue lines because the channels are not closing properly when dialing happen through A2Billing (it seems the dialing scenario including the hangup is not handled properly through A2Billing but I do not have control on this). But when I do dialing from asterisk and using analogue lines, I do not face a trouble because I can write the

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 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

I am unable to get my 9133i to register with my asterisk server. I am including config files below, this a simple test network so there's nothing secret in the config files. I have upgraded the phone to the latest software version (1.4.3) I'm not sure what the problem is. I can call the phone from a softphone, but the 9133i says "no service" on the screen and I can't dial

try "sipsak -M -O desktop -B "foo" -s sip:<user>@<registrar> -H <ip of registrar>" the trick is to specify the "-O desktop" parameter + the "-H <ip of registrar>" parameter. Sipsak fakes the host-header of the registrar so that the Snom thinks it is coming from your Asterisk server, then lets the message through to the

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

The date format may have been an issue, and I had noticed the whois changed the expiration date later to 2018, but dig with my local and with google dns still comes back with no ip. If you addess the gandi dns servers you do get the ip address?? So not sure why it has updated the other dns servers? I have a dyndns.org account, and had mapped an address to the IP and it works find, so the

Hi everybody, I am trying to make up call flow diagrams for for a setup which include ser as a sip proxy/registrar and asteriks as a voicemail server. Is my sequence correct?: UA 1 send an invite to SER. SER forwards this invite to UA2. UA2 sends back a sends back a 100 trying and 180 ringing message. SER forwards these. However UA2 doesnt answer the phone,so what happens then?...is there a

I have been given the following settings for connecting to a voip provider. The names of the fields match my snom phone, and when configured, the phone both makes and recives phonecalls without issue. I am trying to put the same values in asterisk, but there seems to be one field that doesn't seem to exist in asterisk - that of outbound proxy all suggestions welcome SIP headings account

OK. I'm getting out the fireproof suit because it's coming and my hackles have been raised by a number of comments on the list of late. Disclaimer: No disrespect intended to the individuals of any *specific* thread. I'm a little frustrated over energy wasted on pedantic top/bottom posting crap rather than understanding the technology and industry best-practices which have been

Hello- I am currently testing with a carrier that seems to be having some trouble around toll-free (800 number) access. While a problem, its the resulting behavior that I'm finding disconcerting. When I dial an 800#, I get the following response: -- Executing Macro("SIP/2700-e10b", "carrier-out|18005558355|70|r") in new stack -- Executing

On 2/12/19 11:49 PM, Paul R. Ganci wrote: > > On 2/12/19 10:55 PM, Alice Wonder wrote: >> DNSSEC keys do not expire. Signatures do expire. How long a signature >> is good for depends upon the software generating the signature, some >> lets you specify. ldns I believe defaults to 60 days but I am not sure. >> >> The keys are in DNSSKEY records that are signed