Elastix 3.0 and ISDN lines

fhilliph

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Good day,

My company has been using elastix VoIP server for the last couple of years. I am busy upgrading the server to elastix 3.0 but am stuck at the trunk setups.

Previous version of elastix did not require a username to be set by PEER Details but version 3.0 does?

Is there a way around this or not?

Below are examples:

OLD CONFIG

PEER Details:
disallow=all
allow=alaw
context=from-trunk
host=10.200.221.251
qualify=yes
insecure=port
type=peer

USER Details:
context=from-trunk
type=user

There was no required Username and secret in the old config but the v3.0 config requires it.
 
You really shouldn't be trying to circumvent security put in place to protect you.

Also I'm a little confused as to the title and the question. You state ISDN lines, but are asking questions about SIP trunks?
 
Last edited:
We use Epygi's connected to ISDN lines. Which is why we don't have a username or password.
Its not that I don't want to insert a username and secret, we don't have that.

This system was inherited from someone else and was always working fine so we didn't bother changing to newer technologies.
 
You should be able to add the username and password on the Epygi device?
 
We would actually like to move away from the old isdn lines but we can only get ADSL up to 4Mbps where we are. I don't think that would be enough to use a sip provider like telfree for instance without having calls drop and cut.

We cannot get leased line yet either.

Office has about 30+ users and at the most about 6 concurrent outgoing calls.
 
I will try to set a username and password on the epygi.

Thanx for the tip
 
If your ADSL is stable, then 6 concurrent calls is possible over ADSL. We've also been testing VoIP over LTE with some good success.
 
Thank you Clarotech, and who in SA would you recommend for a SIP account ?
Reporting and billing is important.
 
I can't seem to log into the epygi's. Usernames and passwords were never recorded and the default ones don't work.
Looks like I'll have to reset them over a weekend and start from scratch.
 
I can't seem to log into the epygi's. Usernames and passwords were never recorded and the default ones don't work.
Looks like I'll have to reset them over a weekend and start from scratch.

Might be a good idea to get the docs ready before you do that.
 
Hi Clarotech,

Sorry for the late response. Had no time yet to reset the Epygi's.
Quick other question.. Will my SIP provider provide me with licenced codec or will I have to install that myself since the IP PBX is in my office? Also we have already got licenced g729 codecs but how do I transfer (reinstall) them on a new machine?
 
Depends on how far the provider is willing to go. Most just ensure their side is covered and you deal with your PBX vendor for licenses and setup. If you bought your licenses from Digium you can re-license them once.

A G.729 key must be re-registered if any of the Ethernet devices in your Asterisk server are changed, added, or removed. The unique G.729 license file which is located in your /var/lib/asterisk/licenses directory is tied to the MAC address of all the Ethernet devices installed in your system. A G.729 key can only be re-registered once without authorization from Digium. Digium must be contacted by phone in order to request authorization to have your G.729 key incremented. Digium reserves the right to deny authorization for having a G.729 key incremented.The silence suppression feature is not available using the codec_g729a.so file. Asterisk will generate output similar to "Dropping extra frame of G.729 since we already have a VAD frame at the end" if the remote end is attempting to use silence suppression with G.729. Consult with the remote end to ensure that silence suppression is not being used. You may have problems using G.729 if the remote end attempts to use silence suppression.

http://kb.digium.com/articles/FAQ/H...RL=/apex/knowledgeProduct?c=G_729&popup=false
 
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