Investigating some call quality issues (0.5 – 1 second dead spots in calls) I took a packet capture of a phone call between two extensions on the same PBX. Since I was capturing from the PBX, I was rather surprised to see Wireshark reporting a huge…
I setup a PBX with asterisk and Elastix as a web GUI. I have all incoming calls set to ring to a ring group of our office personnel in a ringall format. The only issue is the time for Asterisk to pickup and pass the phone call to the VOIP phones to…
I'm looking for advices regarding VoIP products. I need to build an in-house VoIP PBX for one of my company's office, and while I'm fairly sure we will go with Asterisk (still leaving the door open to OpenSIPS), I'm not certain I want to build it…
I'm planning an Asterisk configuration that should record videocalls and then feed it to an application.
From what I've researched, it seems like app_h234m is the way to go. But it's not clear to me what are the hardware requirements for this.
Can…
What software/product do you use to monitor voice quality in your system? Do you use Asterisk manager API for making calls and recording audio? Do you have software to receive quality scores?
I have a nagios setup which ensures that SIP is responsive on my Asterisk server, that's straight forward.
My question is, what kind of possibilities are there that the Asterisk server can actually terminate properly with the termination provider?…
At office I am considering setting up an asterisk server to that we can route calls that come into the office internally so that they reach the correct person and to implement a menu system as well.
I have managed to set up a server so that we can…
We have one phone (123) that rings from time to time, displaying what seems like an internal number (6001), but it's not, because we don't use that number and nothing in that range. When you pick up the phone, you hear a dial tone.
The server runs…
I'm using Asterisk 1.4 and am trying to work out a way to bring people into a conference call. In the ideal scenario two people would be talking and one of them would push some keys, then a phone number and then the three of them would be in a…
I want to create a voip service.I have installed asterisk-1.4 on a dedicated remotely hosted debian lenny distro. I made a sip.conf and extensions.conf so as to place a call between two sip phones(i am using xlite 3.0) installed in some other…
We're a Canadian business on the east coast and will soon be opening up a new call centre in Australia. This new call centre will be handling our graveyard shift and will be overlapping with the current call centre. As such, I'd need to be able to…
I've skimmed google on this but haven't found anything too useful.
Is there a way to use Google Contacts with a HardPhone supported by Asterisk PBX?
Google contacts is exposed thru a MSExchange server
Thanks
We have an Asterisk 1.8.7.0 (the Elastix derivative) switchboard.
Every since a month ago, seemingly out of the blue, the switchboard does not recognise DTMF tones any more from mobile phones.
Testing the switchboard using 7777 works.
Testing the…
I have a couple of SIP phones and use callcentric. I get a lot of junk calls. I'd like to implement the following feature and would like some suggestions on how to do this:
Once I get a junk call, I typically hang up. I want to dial some number…
First off: I'm not sure if this should be on superuser or here.
I have recently built a few Asterisk boxes with OpenVOX FXO/FXS ports little or no trouble.
My current project is building an Asterisk box with SIP trunks. My current employer insisted…