Origination problem

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Origination problem

Postby gnu » Sat Oct 24, 2015 9:59 pm

Hi, I have a DID

my provider has many Origination IP's that means the "traffic from my DID" can come from different public IP's.

I have configured my sip.conf with all those IP, each [context] has qualify = yes, I did that to I monitor the status (if Reachable or Not).

Well, the case is that, when the call comes from an IP that has the status REACHABLE the call pass well. Everything work fine BUT sometimes I see calls coming from the IP that is reported as UNREACHABLE in my asterisk sip show peers, when that happen the calls doesn't pass because it says

Retransmission timeout reached on transmission 5379401-0-3072310280@208.X.X.X



I asked the provider to see if he can configure my DID to send the traffic to my PBX from an specific IP but they didn't respond me.

What can I do?

What if I configure the firewall (iptables) to redirect all the traffic coming from the UNREACHABLE IP to another REACHABLE IP of provider?


* Comments are Welcome!
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Re: Origination problem

Postby david55 » Sun Oct 25, 2015 5:05 am

This is the wrong forum for support questions. I'll put the answer in the right place.
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Re: Origination problem

Postby ambiorixg12 » Sun Oct 25, 2015 6:58 pm

First Unreachable generally means you have qualify=yes, but the peer is ignoring OPTIONS requests.

This usually happen when there is network connectivity issue between your PBX and the remote peer, but not necessary the case.

Related to the SIP Retransmissions issue.

Why does this happen?
For some reason signalling doesn't work as expected between your Asterisk server and the other device. There could be many reasons why this happens.
A NAT device in the signalling path. A misconfigured NAT device is in the signalling path and stops SIP messages.
A firewall that blocks messages or reroutes them wrongly in an attempt to assist in a too clever way.
A SIP middlebox (SBC) that rewrites contact: headers so that we can't reach the other side with our reply or the ACK.
A badly configured SIP proxy that forgets to add record-route headers to make sure that signalling works.
Packet loss. IP and UDP are unreliable transports. If you loose too many packets the retransmits doesn't help and communication is impossible. If this happens with signaling, media would be unusable anyway.

What can you do?
https://wiki.asterisk.org/wiki/display/ ... nsmissions


ask your provider to see if he can configure my DID to send the traffic to my PBX from an specific, IP. That is very good choice
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