Jonas Kellens
2010-May-21 12:41 UTC
[asterisk-users] Hanging up call - no reply to our critical packet
Hello list, I am confronted with the following problem : making a call only leasts for about 30 seconds, then the call is ended. The CLI shows : [May 21 14:31:50] WARNING[25345]: chan_sip.c:1980 retrans_pkt: Maximum retries exceeded on transmission 954539948-5060-2 at 192.168.1.100 for seqno 11 (Critical Response) -- See doc/sip-retransmit.txt. [May 21 14:31:50] WARNING[25345]: chan_sip.c:2002 retrans_pkt: Hanging up call 954539948-5060-2 at 192.168.1.100 - no reply to our critical packet (see doc/sip-retransmit.txt). I read about SIP-packets being retransmitted but in the end not being acknowledged and so Asterisk ends the call. The network is like this : analogue phone -- Grandstream GXW4008 -- Linksys WAG160 -- Asterisk server -- ITSP Is this a NAT-problem in the router ?? Can it be resolved ? Kind regards, Jonas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100521/c4541cc0/attachment.htm
Jonas Kellens
2010-May-21 17:05 UTC
[asterisk-users] Hanging up call - no reply to our critical packet
For those having the same problem, my solution was to upgrade to the newest firmware on the Linksys WAG160. It seemed a NAT-problem because NAT-ting was not correctly handled by the firmware. Jonas. On 05/21/2010 02:41 PM, Jonas Kellens wrote:> Hello list, > > I am confronted with the following problem : > > making a call only leasts for about 30 seconds, then the call is > ended. The CLI shows : > > [May 21 14:31:50] WARNING[25345]: chan_sip.c:1980 retrans_pkt: Maximum > retries exceeded on transmission 954539948-5060-2 at 192.168.1.100 for > seqno 11 (Critical Response) -- See doc/sip-retransmit.txt. > [May 21 14:31:50] WARNING[25345]: chan_sip.c:2002 retrans_pkt: Hanging > up call 954539948-5060-2 at 192.168.1.100 - no reply to our critical > packet (see doc/sip-retransmit.txt). > > I read about SIP-packets being retransmitted but in the end not being > acknowledged and so Asterisk ends the call. > > > The network is like this : > > analogue phone -- Grandstream GXW4008 -- Linksys WAG160 -- Asterisk > server -- ITSP > > > Is this a NAT-problem in the router ?? Can it be resolved ? > > > Kind regards, > > Jonas-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100521/645b1adf/attachment.htm