Multiple SIP trunks not registering simultaneously

Asterisk Asterisk Support
1

Hello,

This is my 5th attempt to ask a question at this forum and await some assistance.

Would appreciate some response…thanks

I have done a complete fresh install of AsteriskNow and tried setting up my VOIP trunks. I have run into a problem where both these trunks are not connecting simultaneously. If trunk A is connected then B is unreachable and if if disable trunk A and connect B (gets connected) later then enable trunk A only to see it unreachable. Both are connecting on port 5060

The machine is behind NAT, I have opened 5060 to 5070 on router, pointing them to my local IP. Also I have RTP ports open.

[code]Retransmitting #7 (NAT) to 195.189.173.27:5060:
REGISTER sip:sip.voipfone.net SIP/2.0
Via: SIP/2.0/UDP ext-ip:5060;branch=z9hG4bK60a26380;rport
Max-Forwards: 70
From: sip:username@sip.voipfone.net;tag=as672033c4
To: sip:username@sip.voipfone.net
Call-ID: 3818306941195c78720079c5331a4ab9@[::1]
CSeq: 245 REGISTER
User-Agent: FPBX-2.11.0(11.7.0)
Expires: 120
Contact: sip:username@ext-ip:5060
Content-Length: 0

REGISTER 10 headers, 0 lines
Reliably Transmitting (NAT) to 195.189.173.27:5060:
REGISTER sip:sip.voipfone.net SIP/2.0
Via: SIP/2.0/UDP ext-ip:5060;branch=z9hG4bK4dd66541;rport
Max-Forwards: 70
From: sip:username@sip.voipfone.net;tag=as4ce317a2
To: sip:username@sip.voipfone.net
Call-ID: 3818306941195c78720079c5331a4ab9@[::1]
CSeq: 246 REGISTER
User-Agent: FPBX-2.11.0(11.7.0)
Expires: 120
Contact: sip:username@ext-ip:5060
Content-Length: 0

[2014-09-05 15:33:03] NOTICE[1738]: chan_sip.c:15099 sip_reg_timeout: – Registration for ‘username@sip.voipfone.net’ timed out, trying again (Attempt #145)
Really destroying SIP dialog ‘3818306941195c78720079c5331a4ab9@[::1]’ Method: REGISTER
Reliably Transmitting (NAT) to 195.189.173.27:5060:
OPTIONS sip:sip.voipfone.net SIP/2.0
Via: SIP/2.0/UDP ext-ip:5060;branch=z9hG4bK6eca5df7;rport
Max-Forwards: 70
From: “Unknown” sip:username@ext-ip;tag=as0537d9d4
To: sip:sip.voipfone.net
Contact: sip:username@ext-ip:5060
Call-ID: 33e06bd96a43e9244c5e2a894df1f816@ext-ip:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.7.0)
Date: Fri, 05 Sep 2014 14:33:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

Reliably Transmitting (NAT) to 204.74.213.5:5060:
OPTIONS sip:sip.callwithus.com SIP/2.0
Via: SIP/2.0/UDP ext-ip:5060;branch=z9hG4bK0da997d5;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@ext-ip;tag=as762c39de
To: sip:sip.callwithus.com
Contact: sip:Unknown@ext-ip:5060
Call-ID: 5c6a575f2a9d331a7958019532b24d86@ext-ip:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.7.0)
Date: Fri, 05 Sep 2014 14:33:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:204.74.213.5:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP ext-ip:5060;branch=z9hG4bK0da997d5;rport=47556
From: “Unknown” sip:Unknown@ext-ip;tag=as762c39de
To: sip:sip.callwithus.com;tag=05fa965d41f3adc51e16f9a7acf1c273.0889
Call-ID: 5c6a575f2a9d331a7958019532b24d86@ext-ip:5060
CSeq: 102 OPTIONS
Server: CWU SIP GW
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘5c6a575f2a9d331a7958019532b24d86@ext-ip:5060’ Method: OPTIONS

[/code]

What could be the issue?

Thanks.

2

The register request is not reaching the host or its reply is not reaching you. There is really nothing more that can be deduced from the information presented.

In particular, there is no evidence of more than one remote peer.

If the two peers are actually just different accounts on the same host, you can have problems because it can’t tell the two registrations apart, but that would still produce a response.

I would guess that the problem lies in your router.

3

David55, I genuinely appreciate your reply. Honestly!

Thanks!

Right, I have a machine that connects to vonage on port 5061, on the same machine I have a another provider callwithus that connects on port 5060. So far so good. The moment I try another voip provider (voipfone) on this same machine it just doesnt connect, it says UNREACHABLE.

To check if I have problem with my install, I did another install on my VM on same network (router). Tried voipfone on the VM and guess what, UNREACHABLE. The best part is, if I disconnect callwithus sitting on 5060 from my main machine, then I can connect voipfone on 5060 under my VM. I have swapped the providers too, voipfone in place of callwithus and viceversa, results are same. Cannot connect untill the 5060 port is set free from either machine.

What could be the problem? :’(

I have draytek 2830 that sits behind a netgear firewall. I have checked several times, all ports are open and forwarded properly.

4

Moreover, if I connect any other account with same provider that is currently connected via 5060, then this second account connects.

If the host is same then I can connect as many peer as I want under 5060, the moment I try a different host on 5060, it doesnt connect. :confused:

For example,

Main Machine:
Vonage: 5061 (Ok)
Callwithus: 5060 (Ok) (Host XXX)

VM
Callwithus2ndaccount: 5060 (OK) (Same host as callwithus)

or even

Main Machine:
Vonage: 5061 (Ok)
Callwithus: 5060 (Ok) (Host XXX)
Callwithus2ndaccount : 5060 (Ok) (Same host as callwithus)

5

Hi Jeet,

try to open all ports/IPs on your firewall for you server and then connect your both SIP peers.

Sohaib Khan

6

This sounds to me like a problem with the VM environment, not Asterisk.