I am using a Polycom 501 SIP phone behind NAT. Asterisk server is public with no NAT... everything works on the Asterisk end just fine EXCEPT that I can never check voice mail After about 30 seconds the call drops with these messagess: [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 320893f1-50c13ba3-78c26164 at 192.168.1.54 for seqno 2 (Critical Response) [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1972 retrans_pkt: Hanging up call 320893f1-50c13ba3-78c26164 at 192.168.1.54 - no reply to our critical packet. It seems to me that the problem is the way Asterisk is handling this "critical packet" -- of course it can not be sent to 192.168.1.54, the phone is at that IP behind a NAT and the Asterisk server is not. I can make any other phone call from this same phone as long as it is not voicemail and I can be on the line for hours with no problem. I am really at a loss here. I have searched a bit and come up with nothing other than blaming the UA. I know the Polycoms dont have the best NAT support but besides this it works problem-free. It's odd I can make a call anywhere else even for hours and not have any issues at all but 30 seconds into a voicemail call it just drops.... app5*CLI> sip show peer 17865221569 app5*CLI> * Name : 17865221569 Secret : <Set> MD5Secret : <Not set> Context : blended-lcr Subscr.Cont. : sla_stations Language : en AMA flags : Unknown Transfer mode: closed CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : 17865221569 VM Extension : 14193016245 LastMsgsSent : 0/0 Call limit : 2 Dynamic : Yes Callerid : "" <CENSORED> MaxCallBR : 256 kbps Expire : 63 Insecure : no Nat : Always ACL : No T38 pt UDPTL : Yes CanReinvite : No PromiscRedir : No User=Phone : Yes Video Support: No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 74.CENSORED.213 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Reg. exten : Def. Username: 17865221569 SIP Options : (none) Codecs : 0x104 (ulaw|g729) Codec Order : (g729:20,ulaw:20) Auto-Framing: No Status : OK (130 ms) Useragent : PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Reg. Contact : sip:17865221569 at 192.168.1.54 app5*CLI> core show version Asterisk 1.4.21.1 built by root @ app5 on a i686 running Linux on 2008-07-09 01:41:43 UTC
I am using a Polycom 501 SIP phone behind NAT. Asterisk server is public with no NAT... everything works on the Asterisk end just fine EXCEPT that I can never check voice mail After about 30 seconds the call drops with these messagess: [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 320893f1-50c13ba3-78c26164 at 192.168.1.54 for seqno 2 (Critical Response) [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1972 retrans_pkt: Hanging up call 320893f1-50c13ba3-78c26164 at 192.168.1.54 - no reply to our critical packet. It seems to me that the problem is the way Asterisk is handling this "critical packet" -- of course it can not be sent to 192.168.1.54, the phone is at that IP behind a NAT and the Asterisk server is not. I can make any other phone call from this same phone as long as it is not voicemail and I can be on the line for hours with no problem. I am really at a loss here. I have searched a bit and come up with nothing other than blaming the UA. I know the Polycoms dont have the best NAT support but besides this it works problem-free. It's odd I can make a call anywhere else even for hours and not have any issues at all but 30 seconds into a voicemail call it just drops.... app5*CLI> sip show peer 17865221569 app5*CLI> * Name : 17865221569 Secret : <Set> MD5Secret : <Not set> Context : blended-lcr Subscr.Cont. : sla_stations Language : en AMA flags : Unknown Transfer mode: closed CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : 17865221569 VM Extension : 14193016245 LastMsgsSent : 0/0 Call limit : 2 Dynamic : Yes Callerid : "" <CENSORED> MaxCallBR : 256 kbps Expire : 63 Insecure : no Nat : Always ACL : No T38 pt UDPTL : Yes CanReinvite : No PromiscRedir : No User=Phone : Yes Video Support: No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 74.CENSORED.213 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Reg. exten : Def. Username: 17865221569 SIP Options : (none) Codecs : 0x104 (ulaw|g729) Codec Order : (g729:20,ulaw:20) Auto-Framing: No Status : OK (130 ms) Useragent : PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Reg. Contact : sip:17865221569 at 192.168.1.54 app5*CLI> core show version Asterisk 1.4.21.1 built by root @ app5 on a i686 running Linux on 2008-07-09 01:41:43 UTC