WARNING: this is an automatic post retrieved from the Asterisk-Users Mailing List, not an authored post
July 21, 2011
Tags: asterisk, call, cell phone rings, chan, retrans, sip, softphone, timeout message
I have succeeded to establish calls using OpenBTS/USRP1/Asterisk. but the
problem is that my cell phone rings, I get 2 way audio but after a few
seconds the call is dropped. In my asterisk log I see this:
[Jul 18 11:25:48] WARNING: chan_sip.c:3622 retrans_pkt: Retransmission
timeout reached on transmission email@example.com for seqno 94 (Critical
Response) — See
Packet timed out after 32000ms with no response
[Jul 18 11:25:48] WARNING: chan_sip.c:3651 retrans_pkt: Hanging up
call firstname.lastname@example.org – no reply to our critical packet (see
In the SIP Debug, I see always 10 Retransmissions of the same “SIP/2.0 200
Ok” message!!! after that the above “Retransmission timeout” message is
Retransmitting #10 (no NAT) to 127.0.0.1:5062:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK53934;received=127.0.0.
CSeq: 94 INVITE
Server: Asterisk PBX 126.96.36.199-1digium1~lucid
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
Supported: replaces, timer
I have established only a one call from my hardphone (connected to OpenBTS)
to my twinkle softphone. but after the call is dropped (T == 32 secondes) by
my softphone and after hanging up my hardphone (T == 60 seconds) I have
received automatically a call from my twinkle softphone!!!
In wireshark trace, I see that OpenBTS is trying to ACK the OK from
Asterisk, but Asterisk doesn’t like it !!!
I have tried to modify the value of the SIP timers, that works only from a
hardphone to a softphone but not from hard to hard. can some one tell us
what’s the definition of t1min and timert1?
Any help will be appreciated.