Sip Trunks

Asterisk Asterisk Support
1

Hi Folks - Hope I am in the right place.

We been running our VoIP through an ISDN line with Verizon. That line was very unreliable and would fail often so we contacted a sip carrier and purchased a number of sip trunks to replace the ISDN line.

We are running Asterisk 1.6.2.9-2 on a Linux server. My question is - the new carrier sent me IP addresses for incoming and outgoing calls but I don’t know what config file to modify.
Can someone give me some details on how to set up our VoIP for the sip trunks - which files to configure? Even if I could follow some samples would be helpful.

Thank you much.

2

You have tons of samples for SIP trunking on this forum or on the internet. Make an effort and browse the existing documentation.

In general you need to:

  • define a peer in sip.conf
  • route calls from/to the sip peer in extensions.conf
3

Additionally, any competent supplier of sip services to PABX users will be able to provide you with information on how to configure sip.conf for, at least an obsolete version of Asterisk, which you can use as a starting point.

There is a problem that they tend to provide over-permissive configurations that use options that have since been renamed.

4

Efforts made - that’s why I asked here.

5

If an effort has been made, I would expect that you tried some configurations and they did not work for you.

What did you try and what did not work?

6

I am not getting incoming calls - I am not asterisk or VoIP experienced so I thought I may see some sample cofigs and try them - I don’t know exactly what these strings mean so manipulating them is hit and miss for me - that’s where I thought I could get some insight here. The string beneath “bandwidth” makes no sense to me - if I could get a brief explanation of what it is doing - that may help me better.

I copied one of the samples that looked as close to my confs as possible and replaced my number ( the xxxxxxxxxx) with the number in the sample.

This is my extensions.conf file …

[bandwidth]
exten => _+1.,1,Set(DID=${EXTEN:2})
exten => _+1.,n,Goto(fromtrunk,${DID},1)

[fromtrunk]
exten => xxxxxxxxxx,1,Verbose(1|Unrouted call handler)
exten => xxxxxxxxxx,n,Answer()
exten => xxxxxxxxxx,n,Wait(1)
exten => xxxxxxxxxx,n,Dial(SIP/7xxxx,60,tT)
exten => xxxxxxxxxx,n,Goto(phones,7xxxx,2)
exten => xxxxxxxxxx,n,Hangup()

I modified my SIP.conf file with the IP addresses supplied by bandwidth - that config file seems to be more straight foward and common so I am pretty sure that is ok.

I will not be disappointed if this kind of help is not what this forum can provide - I was just looking for some direction.