I am using a SIP provider that has provided me with a username like:
+1122233344@aaa.bb.com (Note this is only the username part)
And has a numerical password. My Register string looks something like…
I successfully set up an asterisk server. When people call my asterisk server via PSTN, the server will place another PSTN call to my phone at 33344455555. When I receive the call, my phone shows that I'm receiving a call from 4169998888, which is…
Timing is very important for certain kinds of applications in Asterisk. If DAHDI is the timing source, the dahdi_test command can be used to check the timing provided by the DAHDI kernel module. If dahdi_test returns exclusively measurements above…
I have an Asterisk 1.8 phone switch with a Digium T1 card. It runs using 5ESS PRI through our present phone provider without a problem. However, we're contemplating switching to Time Warner's fiber service (not TWTelecom) and then it fails with ISDN…
I've recently moved to a Cisco 881 router for my WAN link. I was previously using a Cisco Linksys WAG320N as my modem/router/WiFi AP/NAT firewall. The WAG320N is now running in bridged mode, so it's simply acting as a modem with one of it's LAN…
I want a caller to be able to phone into asterisk from a mobile phone or a pay phone and then enter a special PIN and allow them to dial another number.
This could allow for significant savings on long distance calls.
Is there a build in module or…
I am setting up a new server using Asterisk 1.8.11-certified4. In testing, we're seeing that agents dynamically logged into the queue will receive a second queue call as a call-waiting when call-limit is set to 0.
Since the agents in question are…
I have setup an Elastix box (asterisk/freepbx based) and added a trunk for an external VoIP provider.
When I get to the Asterisk command line interface and type sip show registry I always get the same output, State = Request Sent.
So, since I can't…
I am working with asterisk (Asterisk 1.8.11.0), freepbx (2.x) and was building a dialplan (extension_custom.conf).
I wanted to get caller Name from database which i have stored for example.
CLI> database show cidname
/cidname/XXXXXXXXXX …
I installed VMware and then Ubuntu. I am running asterisk on top of it.
I started the asterisk server successfully by following a tutorial. After making few changes into the sip.conf, I want to reload the the process so it pick up the changes. So I…
I'm running into a funny little issue with Asterisk 10.3, but it seems to be applicable to 10.4 as well.
The server running Asterisk was relocated from a VPS to dedicated hardware, and now only 1 of several SIP peers can connect properly.
SIP peers…
I'm having a problem with a debian server that I thought was due to bad RAM, but is persisting.
It's a Dell Poweredge 6800 with two dual-core 3.6GHZ Xeon processors and 5GB of DDR2 ECC 333.
I've got a single 73GB SCSI Drive.
I'm working it to death…
I'm in pain.
We are moving to a SIP based VOIP system and for whatever reason, we could not get our hosted Asterisk solution to work with our Sonicwall. Our VOIP provider gave up and is recommending an open source vendor, pfSense.
A little…
Basically, I want to do the same thing that Google Voice does. I forward my calls to a DID number that rings my Asterisk server via IAX2, which, if it detects the call has been forwarded, sends it to voicemail. Otherwise, if the call hasn't been…