Yes ....
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Thursday, March 11, 2004 10:47 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call You do have : nat_enable: "1" nat_received_processing: "1" On the Ciscos? AstGrp wrote: >I am having a similar problem... I get the same message, but inbound >calls can go through.... This is only Cisco phones that are behind NAT. >I have tried your recommendations from below, but still no luck.. User >can make outbound calls, just can't receive any. Any ideas would be >greatly appreciated.. I even tried to change the timeout value in >chan_sip, but it just waits longer to fail.. Just dosen't seem to want >to communicate... > >Thanks, > >gcc > >-----Original Message----- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] On Behalf Of John >Bittner Posted At: Tuesday, March 02, 2004 11:46 PM Posted To: Asterisk >User Group >Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum >retries exceeded on call >Subject: RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum >retries exceeded on call > > >Are you using Cisco phones. ? > >I had this issue with my cisco phones. I didn't had any issues with >dropped calls. All I did to fix this was set a prefered_codex and set >proxy_register to 0. > >I hope this helps. > >John Bittner >Simlab.net > > > > >>-----Original Message----- >>From: [EMAIL PROTECTED] >>[mailto:[EMAIL PROTECTED] On Behalf Of dkwok >>Sent: Wednesday, March 03, 2004 7:04 AM >>To: [EMAIL PROTECTED] >>Subject: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum >>retries exceeded on call >> >>*CLI> Mar 3 12:55:05 WARNING[1150495040]: chan_sip.c:495 >>retrans_pkt: >>Maximum retries exceeded on call >>[EMAIL PROTECTED] for seqno 102 (Request) >> >>This has been brought up in the previous post but it does not seem to >>have an answer for it so far. >> >>I cvs the stable v1.0 this morning after compiling and installing I >>have calls drop 1 minutes into the connection with the above message. >> >>If anyone has any idea of this occurrence. >> >>I have set up sip.conf: >> >>canreinvite=no >> >>-- >>David Kwok >>Tel: 612 99292086 ext 1002 >>Iaxtel/FWD # 17001813482 ext 1002 >> >> >> > >_______________________________________________ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users