Aststerisk 1.6 is hanging up automatically after 3 seconds


I’m new to this forum and of course new in the matter of Asterisk. My problem is that every incomming call is breaking down after 3 seconds.

On the Asterisk console appears following message:

[Jul 27 13:48:35] WARNING[6306]: chan_sip.c:3785 retrans_pkt: Maximum retries exceeded on transmission 71bfa420-1428-122e-a9b8-001b245d6c13 for seqno 134000911 (Critical Response) -- See doc/sip-retransmit.txt. [Jul 27 13:48:35] WARNING[6306]: chan_sip.c:3812 retrans_pkt: Hanging up call 71bfa420-1428-122e-a9b8-001b245d6c13 - no reply to our critical packet (see doc/sip-retransmit.txt). == Spawn extension (ankommend, 0, 1) exited non-zero on 'SIP/xpirio-00000008'

From the SIP Debug i get following message:

Firstly i thought it has something to do with the ports, i forwarded port 5060 and the configured RTP Port Range in rtp.conf, but it doesn’t help.

I should say that my Router is an IPFire OS installed in a Xen DomU, and Asterisk is connected to the “Blue Zone”. (Asterisk 1.6 is installed on a Debian 5.0 DomU on the same XEN Server)

I would be grateful if you could help me to solve this problem!


This is a port 5060 routing problem. Possibly a failure to configure NAT features correctly. RTP isn’t involved yet.

A more complete trace would make it clearer which packet wasn’t getting through, but I suspect it is the OK to an incoming INVITE.

IP telephony and virtual machines don’t mix well, although the failure will be poor audio quality, rather than call setup failures. This will apply to routers used for VoIP, as well as switches.

3 seconds seems too short for a “no response to critical packet” type error.

Thanks for your reply!

Ok if it is a Port 5060 NAT problem, how can a make a complete trace?

My Asterisk has only 10 SIP Phones connected, and its possible to make 2 calls to the sip provider at the same time, i think the performace of my XEN Server is more than enough ( AMD Phenom X3 Black Edition 3x 2,4 GHZ, 12GB RAM (2GB Asterisk DomU))


sip set debug on

The problem with virtual machines is that they receive slices of time. If there is a lot of activity, those slices can at irregular and extended intervals. That is OK if long term average power is good, but not if you need short term responsiveness.

ok I’ll did the sip debug again and I got this message

[Sep 19 19:30:51] WARNING[20127]: chan_sip.c:3831 retrans_pkt: Maximum retries exceeded on transmission 709b03d9-3eb6-122e-0ba7-001b245d6c13 for seqno 2122650 (Critical Response) -- See doc/sip-retransmit.txt. [Sep 19 19:30:51] WARNING[20127]: chan_sip.c:3858 retrans_pkt: Hanging up call 709b03d9-3eb6-122e-0ba7-001b245d6c13 - no reply to our critical packet (see doc/sip-retransmit.txt). Scheduling destruction of SIP dialog '55f202b17c55bc111e6d7f76375ae0a5@' in 32000 ms (Method: INVITE) set_destination: Parsing <sip:42@;transport=udp> for address/port to send to set_destination: set destination to, port 5060 Reliably Transmitting (no NAT) to BYE sip:42@;transport=udp SIP/2.0 Via: SIP/2.0/UDP;branch=z9hG4bK57271960;rport Max-Forwards: 70 From: "<112345666>" <sip:112345666@>;tag=as5f1d4e9d To: <sip:42@;transport=udp>;tag=001873567fa600070deef539-4463dcb1 Call-ID: 55f202b17c55bc111e6d7f76375ae0a5@ CSeq: 103 BYE User-Agent: Asterisk PBX X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0

Now I installed Asterisk on a completely new computer without xen, but it makes no difference :frowning:


It is definitely a NAT issue.

Check that these things are setup:

externip = {your statis ip}
localnet = (assuming that is your network and netmask)

then make sure you have this set for all of your extensions in Sip.conf or realtime db


is a fixed ip address mandatory? because i don't have one :frowning:

is a fixed ip address mandatory? because i don’t have one :frowning:

You have to configure a stun server if you don’t have a fixed address.

At the very least you need a stable address, i.e. one that won’t keep changing within a single session.

Can you try and define DMZ on your router? What kind of router do you have?

Instead of externip you may use externhost, but you have to use dyndns then.

I use the IP Fire Router Software in a separate Xen Domain (
At IP Fire the DMZ is called orange zone. (my Asterisk is connected to the orange zone). I’m using IPFire because its free, and it supports QoS.

I already have DynDNS, but how can i use it in my configuration?

I’ve found something on the Internet about Asterisk and DynDNS, are these settings correct?

externhost=<> externrefresh=10 nat=yes localnet=

i’ve now added these settings from my last post to my sip.conf but it makes no difference, i’ve got the following message after few seconds and the call is terminated

[Oct 2 11:57:15] WARNING[3251]: chan_sip.c:3831 retrans_pkt: Maximum retries exceeded on transmission 1b8d827c-48ae-122e-4c81-001b245d6c13 for seqno 2670617 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 2 11:57:15] WARNING[3251]: chan_sip.c:3858 retrans_pkt: Hanging up call 1b8d827c-48ae-122e-4c81-001b245d6c13 - no reply to our critical packet (see doc/sip-retransmit.txt). == Spawn extension (ankommend, 0, 1) exited non-zero on 'SIP/xpirio-00000000'

it drives me insane now!!!

It works!!!

i’ve added the external ip and the localnet to the sip provider settings in the sip.conf. I now added it to the general settings and it works.

thanks for your help!

It works for me too! :smiley:

Althought I had slight different problem.
Asterisk wouldn’t register to sip provider with the same message as mentioned above:

2013-12-11 08:55:14 WARNING[140382713419520]: chan_sip.c:1682 retrans_pkt: Maximum retries exceeded on transmission for seqno 125 (Critical Request) SIP Timer T1 : 500
2013-12-11 08:55:14 NOTICE[140382713419520]: chan_sip.c:8292 sip_reg_timeout: – Registration for '’ timed out, trying again (Attempt #24)

Adding externip and localnet to sip.conf solved this. :laughing:
I thought I’ll share my experience, when somebody faces same issue like me.
Thank you.

Your newest member,
Aas. :wink: