One way of keeping a little black book for all your telephone contacts is through-
Under trunks configure the following: Outbound Caller ID: This is your national dial in number PEER Details: type=friend disallow=all allow=alaw host=sip.clubtelco.com username=09xxxxxx fromuser=09xxxxxx authuser=09xxxxxx secret=Password port=5060 maxexpirey=240 defaultexpirey=240 outboundproxy=sip.clubtelco.com fromdomain=sip.clubtelco.com context=from-trunk qualify=yes insecure=invite canreinvite=no Register String: 09XXXXXX:Password@sip.clubtelco.com/09XXXXXX Note: I didn’t have to enter anything under the user details section. Elastix version is 1.6-12 and freepbx.
We normally use ivox as our voip provider where it automatically relays dtmf tones over the trunk. We now use a variety of providers where they all don’t automatically support dtmf. To get it working you need to add the following to your outbound trunk settings. dtmfmode=inband Basically this adds the dtmf tones to.
By default, Asterisk will store voice messages in the spool folder, at /var/spool/asterisk/voicemail/<context>/<mailbox> You many note there are .WAV and .wav files of the same name e.g. unavail.wav and unavail.WAV with the .WAV file being larger – about 10 times the size. These are different quality of recordings. Here
Note to unlock menu items on the phone that are locked, press-
If you are using a Linksys SPA942 and would like to enable viewing of other users status, or be able to press a button to call that extension here is how to configure Make sure you have the latest phone firmware 6.1.5(a). Connect to the web interface on the SPA942 and under the info tab.
I setup a US number with anveo $2.49 setup and $1.49 per month for a DID. Online chat support was helpful and they have a comprehensive call routing system including being able to accept faxes on the DID, setup voicemail etc. Call quality is good from Australia to US numbers. Incoming calls worked fine on.
Due to attempted hacks on VOIP boxes we need to increase security by allowing SIP access only from the relevant IPs using iptables firewall In the case below the Asterisk VOIP server is sitting behind a NAT firewall and has the relevant config set for this. I found upon enabling the linux firewall that while.
If running your asterisk server behind a firewall you may experience these symptoms: internal users can call mobile extensions configured with follow me and the call goes through and can be heard with no problem. If you have a rule configured so that external callers after a timeout calling internal extensions are then routed to.
After doing a backup of asterisk from one system, and restoring to a fresh build, you may find that when someone goes to voicemail, there is a seconds silence then it drops out. If you use *97 and try to record a message, it immediately drops back to the options to save the message without.