Cisco 7911 SIP Retransmission in Asterisk 1.8

Asterisk Asterisk Support
1

Hello

When migrating from asterisk 1.6 to Asterisk 1.8 my Cisco 7911 Phones hungup the calls in about 32 seconds. Only when calling between Cisco 7911 phones. Calls involving any softphone are 100% OK

In the console I could see:

[Jan 12 11:58:10] WARNING[25661]: chan_sip.c:3622 retrans_pkt: Retransmission timeout reached on transmission 00260bd7-88330025-61992622-2d11eb65@10.1
.13.100 for seqno 102 (Critical Response) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 32000ms with no response
[Jan 12 11:58:10] WARNING[25661]: chan_sip.c:3651 retrans_pkt: Hanging up call 00260bd7-88330025-61992622-2d11eb65@10.1.13.100 - no reply to our criti
cal packet (see wiki.asterisk.org/wiki/display/ … nsmissions).

I tried several 1.8 releases whith unsuccessfull results. In fact, I had to downgrade to asterisk 1.6 where everything works fine.

I’m not using NAT

Does anybody can help me?

Best regards
Ramón Jiménez

2

What was the packet that was retransmitted? (sip set debug on)

3

It’s an INVITE 200 OK. The best approach of solution I fund in internet was

wiki.asterisk.org/wiki/display/ … nsmissions

But none of the solutions were effective.

4

If it is the original INVITE, it is a firewall, or NAT problem.

If it is a re-invite, it may be broken firmware in the phone.