Asterisk & Voipfone PBX

DataMetrics_Ltd · Joined: 13 Feb 2006 Posts: 20 Location: Norwich

Hi,

I currently run an Asterisk server and have a number of PBX lines provided by Voipfone. I use the PBX lines from Voipfone to provide direct dial in for sales, accounts, support etc.

Everything has been working well, but for some reason the SIP pears keep saying 'Request sent', and when this happens no calls come through. As soon as I tell the server to re-read its configs all of the sip peers re-register and calls can come in once more.

The server has it's own public IP address, and therefore does not have any NAT routing problems. However the machine does have two network cards, one for allowing my local network to talk to the machine (SIP phones on localnetwork) and the other provides access to the public IP.

Does anyone have any suggestions on what the problem could be, or how i would go about fixing this?

Thanks in advance,

Ross

DataMetrics_Ltd · Joined: 13 Feb 2006 Posts: 20 Location: Norwich

Hi,

I have been looking into this problem further, and I am left wondering if the problem is caused by the username for the virtual PBX.

Due to regular expresion matching used by asterisk to match its sip responses, could the username 888888*200 etc actualy mach all other extensions such as 888888*201 ect?

I have tested this theory by removing all of my other PBX extensions from Asterisk and keeping my normal account number (the one without the *) and just one other extension. It seams that the username without the '*' keeps registered however the username with the '*' does not.

I realy need to have at least 2 peers in my Asterisk server, to determin where a call is headed (ie sales, support etc).

Is it possible to change the usernames on my pbx account so that I can do this, or sign up for another account and have the required numbers moved to that account?

Ross

DataMetrics_Ltd · Joined: 13 Feb 2006 Posts: 20 Location: Norwich

Hi,

After doing some digging it appears that Asterisk can only support one trunk from a single host. This is due to how it matches the peer's sip messages to its internal records.

This is a problem for me as I need to have multiple trunks.

Is there any other host I can register to? (i.e. not just voipfone.co.uk)

I'm guessing that you have more than one registration server that handles the requests, so would it be possible for me to point an additional trunk to this host. If so can you please let me know ASAP.

Regards,

Ross

markiebrown · Joined: 13 Mar 2006 Posts: 54

Did u ever get a reply to your query?
Mark

MrMan · Joined: 04 Feb 2006 Posts: 161

There is another IP for a new server, but everyone will be using that soon, so it looks like you have a bit of a problem, as do i!

I have only just worked out why some of the trunks on my Aterisk keep going offline, because of their being more than one account on same host, you helped me with that Very Happy

markiebrown · Joined: 13 Mar 2006 Posts: 54

To be honest I have had exactly the same problem with an Asterisk server that only has one line registered.. so not entirely convinced that thats the only reason we keep losing registration.
But we live in hope that things will get better when they finally introduce IAX. Lets hope its not something they will never introduce. In all honesty the lack of IAX is the reason I'm not pushing to use VF for my clients.
Mark Smile

arcascomp · Joined: 20 Mar 2006 Posts: 41

On the website you can change the name of an extension - not sure if this changes the details required for logging in via Asterisk. Worth a try?

suburban74 · Joined: 28 Feb 2006 Posts: 60

hi chaps

was there any outcome to this.. i am testing Trixbox / AAH and also would like multiple trunks to reflect seperate lines of business.

with regards tp IAX will this work with dynamic dns (i don't have a static ip)

thanks

najemhasan · Joined: 11 Oct 2006 Posts: 24

We set up our trixbox behind a DMZ using a Linksys router (WAG354G v2) and it works exceptionally well.

One thing we found was that by upgrading freepbx to the latest version many of our issues were resolved - however, upgrading trixbox broke most of what we fixed so had to revert to 1.2.3

We have now become quite proficient in asterisk/trixbox - if you can post specific questions I can try and help...

Just FYI, our trixbox has three NIC's, one zaptel card and we seem to have no issue with routing. Early on in deployment we found that the default route was often the cause of any connectivity issue...

If anyone is interested, we now provide setup and maintenance contracts on trixboxes for a reasonable fee - pm me if you would like more details.

Naj

Customer Services · Site Admin Joined: 24 Aug 2006 Posts: 233

najemhasan · Joined: 11 Oct 2006 Posts: 24

Lol - thanks for the reply!

Perhaps you should update the thread advising how we set up the trunk in the first place:

http://www.voipfoneuserforum.com/viewtop ic.php?t=1013

Thats what we followed Smile

N

Bins · Joined: 18 Mar 2006 Posts: 25

look on the main site. there is a specific reference to asterisk, and, if logged in to your account, it will offer something yoy could just cut/paste