Call ends by itself [SOLVED BY VPN]

General Help
1

Hi! I’m having a problem with my FreePBX… I receive calls from a PSTN line, in a Sipura SPA3000 trunked (http://www.freepbx.org/support/documentation/howtos/howto-linksys-spa-3102-sipura-spa-3000-freepbx) and I answer with a cisco 7940… Here is my problem… After a few seconds, my calls ends by them self… Here is the end of the CLI windows (with vvv verbosity) :

    -- Executing [[email protected]:42] Dial("SIP/ottawa-00000068", "SIP/205,15,tr") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/205
    -- SIP/205-00000069 is ringing
    -- SIP/205-00000069 is ringing
    -- SIP/205-00000069 answered SIP/ottawa-00000068
[2012-11-28 13:58:07] WARNING[3207]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission [email protected] for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[2012-11-28 13:58:07] WARNING[3207]: chan_sip.c:3670 retrans_pkt: Hanging up call [email protected] - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
    -- Executing [[email protected]:1] Macro("SIP/ottawa-00000068", "hangupcall,") in new stack
    -- Executing [[email protected]:1] GotoIf("SIP/ottawa-00000068", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [[email protected]:3] ExecIf("SIP/ottawa-00000068", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [[email protected]:4] Hangup("SIP/ottawa-00000068", "") in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/ottawa-00000068' in macro 'hangupcall'
  == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/ottawa-00000068'
  == Spawn extension (macro-dial-one, s, 42) exited non-zero on 'SIP/ottawa-00000068' in macro 'dial-one'
  == Spawn extension (macro-exten-vm, s, 14) exited non-zero on 'SIP/ottawa-00000068' in macro 'exten-vm'
  == Spawn extension (from-did-direct, 205, 2) exited non-zero on 'SIP/ottawa-00000068'

I’m not an expert with Asterisk… I can’t understand everything from the CLI…

Thank’s!

2 
Retransmission timeout reached on transmission [email protected] for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response

You have a network/NAT issue. Asterisk can’t reach one of the devices to make the transfer.

3

Packets always time out after 6400… Do you have any suggestion on where to look for? The place where this phone is install may have some internet issues… but I’m not allowed to play in this internet… Do you think I should change a port? Use something weird like 2070 instead of 5060? I’ve two phones their… One Cisco 7940 (5060) and one SPA3000 (5061)…

4

I would setup a VPN

5

Thank’s… I’ll see what I can do…

6

I finally solve my problem by adding a VPN!

I was already having a site-to-site VPN between the server site and one of my remotes, using DD-WRT routers OpenVPN so I just bought a new router (Asus RT-N16, which I recommend! Good router! Would be interesting to see if we can install FreePBX on it…)

Thank’s for your help!

Frederic

7

Hey guys I’m having this issue as well and I’m not sure why. I have openvpn running to the network where the asterisk server is. The OpenVPN connection is on the “loc” network so it has full access to the network. I don’t know why it would fail and calls would drop. Could you please point me in the right direction? Any logs I can post?

Thanks

8