-- Executing [EMAIL PROTECTED]:2] VoiceMailMain("SIP/17865221569-b6b03f60", "3523782778|s") in new stack -- <SIP/17865221569-b6b03f60> Playing 'vm-youhave' (language 'en') app5*CLI> <--- SIP read from 74.170.252.213:5060 ---> ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bKaf5e8319A2DD152C From: "17865221569" <sip:[EMAIL PROTECTED]>;tag=329CAFE3-451838A4 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as530e3156 CSeq: 2 ACK Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Proxy-Authorization: Digest username="17865221569", realm="netjdn.com", nonce="01c73ede", uri="sip:[EMAIL PROTECTED]:5060;user=phone", response="7b07733275a9401cc5d8bbdfcd4028b4", algorithm=MD5 Max-Forwards: 70 Content-Length: 0
<-------------> --- (12 headers 0 lines) --- -- <SIP/17865221569-b6b03f60> Playing 'digits/1' (language 'en') Retransmitting #2 (NAT) to 74.170.252.213:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bK36ec371e46AEDA45;received=74.170.252.213 From: "17865221569" <sip:[EMAIL PROTECTED]>;tag=329CAFE3-451838A4 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as530e3156 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: HardenedSipServer-4.x Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 291 v=0 o=root 26803 26803 IN IP4 74.124.208.137 s=session c=IN IP4 74.124.208.137 t=0 0 m=audio 10624 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- app5*CLI> <--- SIP read from 74.170.252.213:5060 ---> ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bKaf5e8319A2DD152C From: "17865221569" <sip:[EMAIL PROTECTED]>;tag=329CAFE3-451838A4 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as530e3156 CSeq: 2 ACK Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Proxy-Authorization: Digest username="17865221569", realm="netjdn.com", nonce="01c73ede", uri="sip:[EMAIL PROTECTED]:5060;user=phone", response="7b07733275a9401cc5d8bbdfcd4028b4", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- -- <SIP/17865221569-b6b03f60> Playing 'vm-INBOX' (language 'en') -- <SIP/17865221569-b6b03f60> Playing 'vm-and' (language 'en') -- <SIP/17865221569-b6b03f60> Playing 'digits/8' (language 'en') app5*CLI> <--- SIP read from 74.170.252.213:5060 ---> REGISTER sip:sip02.netjdn.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bKe9cfe68bDC7AD066 From: "17865221569" <sip:[EMAIL PROTECTED]>;tag=256F3E05-5ECA57DE To: <sip:[EMAIL PROTECTED]> CSeq: 6105 REGISTER Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Authorization: Digest username="17865221569", realm="netjdn.com", nonce="3a435c0d", uri="sip:sip02.netjdn.com:5060", response="9e6d6128a5e6e4508dca68b29c2c277c", algorithm=MD5 Max-Forwards: 70 Expires: 30 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 74.170.252.213 : 5060 (NAT) <--- Transmitting (NAT) to 74.170.252.213:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bKe9cfe68bDC7AD066;received=74.170.252.213 From: "17865221569" <sip:[EMAIL PROTECTED]>;tag=256F3E05-5ECA57DE To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 6105 REGISTER User-Agent: HardenedSipServer-4.x Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 <------------> app5*CLI> <--- Transmitting (NAT) to 74.170.252.213:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bKe9cfe68bDC7AD066;received=74.170.252.213 From: "17865221569" <sip:[EMAIL PROTECTED]>;tag=256F3E05-5ECA57DE To: <sip:[EMAIL PROTECTED]>;tag=as5e3cef8a Call-ID: [EMAIL PROTECTED] CSeq: 6105 REGISTER User-Agent: HardenedSipServer-4.x Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: <sip:[EMAIL PROTECTED]>;expires=60 Date: Wed, 08 Oct 2008 20:14:48 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: REGISTER) -- <SIP/17865221569-b6b03f60> Playing 'vm-Old' (language 'en') Retransmitting #3 (NAT) to 74.170.252.213:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bK36ec371e46AEDA45;received=74.170.252.213 From: "17865221569" <sip:[EMAIL PROTECTED]>;tag=329CAFE3-451838A4 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as530e3156 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: HardenedSipServer-4.x Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 291 v=0 o=root 26803 26803 IN IP4 74.124.208.137 s=session c=IN IP4 74.124.208.137 t=0 0 m=audio 10624 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- app5*CLI> <--- SIP read from 74.170.252.213:5060 ---> ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bKaf5e8319A2DD152C From: "17865221569" <sip:[EMAIL PROTECTED]>;tag=329CAFE3-451838A4 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as530e3156 CSeq: 2 ACK Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Proxy-Authorization: Digest username="17865221569", realm="netjdn.com", nonce="01c73ede", uri="sip:[EMAIL PROTECTED]:5060;user=phone", response="7b07733275a9401cc5d8bbdfcd4028b4", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- app5*CLI> <--- SIP read from 74.170.252.213:5060 ---> BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bK5ffcb43bD93CC4D6 From: "17865221569" <sip:[EMAIL PROTECTED]>;tag=43A660C8-2E49305 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as04d385ce CSeq: 3 BYE Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Proxy-Authorization: Digest username="17865221569", realm="netjdn.com", nonce="3e862d54", uri="sip:[EMAIL PROTECTED]:5060;user=phone", response="e741febb8b521e03c0cd813820cded12", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- -- <SIP/17865221569-b6b03f60> Playing 'vm-messages' (language 'en') -- <SIP/17865221569-b6b03f60> Playing 'vm-onefor' (language 'en') -- <SIP/17865221569-b6b03f60> Playing 'vm-INBOX' (language 'en') -- <SIP/17865221569-b6b03f60> Playing 'vm-messages' (language 'en') -- <SIP/17865221569-b6b03f60> Playing 'vm-opts' (language 'en') Retransmitting #4 (NAT) to 74.170.252.213:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bK36ec371e46AEDA45;received=74.170.252.213 From: "17865221569" <sip:[EMAIL PROTECTED]>;tag=329CAFE3-451838A4 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as530e3156 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: HardenedSipServer-4.x Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 291 v=0 o=root 26803 26803 IN IP4 74.124.208.137 s=session c=IN IP4 74.124.208.137 t=0 0 m=audio 10624 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- app5*CLI> <--- SIP read from 74.170.252.213:5060 ---> ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bKaf5e8319A2DD152C From: "17865221569" <sip:[EMAIL PROTECTED]>;tag=329CAFE3-451838A4 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as530e3156 CSeq: 2 ACK Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Proxy-Authorization: Digest username="17865221569", realm="netjdn.com", nonce="01c73ede", uri="sip:[EMAIL PROTECTED]:5060;user=phone", response="7b07733275a9401cc5d8bbdfcd4028b4", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- app5*CLI> <--- SIP read from 74.170.252.213:5060 ---> BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bK5ffcb43bD93CC4D6 From: "17865221569" <sip:[EMAIL PROTECTED]>;tag=43A660C8-2E49305 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as04d385ce CSeq: 3 BYE Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Proxy-Authorization: Digest username="17865221569", realm="netjdn.com", nonce="3e862d54", uri="sip:[EMAIL PROTECTED]:5060;user=phone", response="e741febb8b521e03c0cd813820cded12", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Retransmitting #5 (NAT) to 74.170.252.213:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bK36ec371e46AEDA45;received=74.170.252.213 From: "17865221569" <sip:[EMAIL PROTECTED]>;tag=329CAFE3-451838A4 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as530e3156 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: HardenedSipServer-4.x Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 291 v=0 o=root 26803 26803 IN IP4 74.124.208.137 s=session c=IN IP4 74.124.208.137 t=0 0 m=audio 10624 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- app5*CLI> <--- SIP read from 74.170.252.213:5060 ---> ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bKaf5e8319A2DD152C From: "17865221569" <sip:[EMAIL PROTECTED]>;tag=329CAFE3-451838A4 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as530e3156 CSeq: 2 ACK Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Proxy-Authorization: Digest username="17865221569", realm="netjdn.com", nonce="01c73ede", uri="sip:[EMAIL PROTECTED]:5060;user=phone", response="7b07733275a9401cc5d8bbdfcd4028b4", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- app5*CLI> <--- SIP read from 74.170.252.213:5060 ---> BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bK5ffcb43bD93CC4D6 From: "17865221569" <sip:[EMAIL PROTECTED]>;tag=43A660C8-2E49305 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as04d385ce CSeq: 3 BYE Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Proxy-Authorization: Digest username="17865221569", realm="netjdn.com", nonce="3e862d54", uri="sip:[EMAIL PROTECTED]:5060;user=phone", response="e741febb8b521e03c0cd813820cded12", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 8640 ms (Method: NOTIFY) Reliably Transmitting (NAT) to 74.170.252.213:5060: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 74.124.208.137:5060;branch=z9hG4bK54fdc202;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as42840967 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: HardenedSipServer-4.x Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: no Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 0/0 (0/0) --- app5*CLI> <--- SIP read from 74.170.252.213:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 74.124.208.137:5060;branch=z9hG4bK54fdc202;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as42840967 To: <sip:[EMAIL PROTECTED]>;tag=DED38D10-3F2880AD CSeq: 102 NOTIFY Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '[EMAIL PROTECTED]' Method: NOTIFY -- <SIP/17865221569-b6b03f60> Playing 'vm-helpexit' (language 'en') Retransmitting #6 (NAT) to 74.170.252.213:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bK36ec371e46AEDA45;received=74.170.252.213 From: "17865221569" <sip:[EMAIL PROTECTED]>;tag=329CAFE3-451838A4 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as530e3156 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: HardenedSipServer-4.x Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 291 v=0 o=root 26803 26803 IN IP4 74.124.208.137 s=session c=IN IP4 74.124.208.137 t=0 0 m=audio 10624 RTP/AVP 18 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- app5*CLI> <--- SIP read from 74.170.252.213:5060 ---> ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bKaf5e8319A2DD152C From: "17865221569" <sip:[EMAIL PROTECTED]>;tag=329CAFE3-451838A4 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as530e3156 CSeq: 2 ACK Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Proxy-Authorization: Digest username="17865221569", realm="netjdn.com", nonce="01c73ede", uri="sip:[EMAIL PROTECTED]:5060;user=phone", response="7b07733275a9401cc5d8bbdfcd4028b4", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- [Oct 8 16:15:04] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 2 (Critical Response) [Oct 8 16:15:04] WARNING[26819]: chan_sip.c:1972 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. == Spawn extension (macro-vmlogin, s, 2) exited non-zero on 'SIP/17865221569-b6b03f60' in macro 'vmlogin' == Spawn extension (macro-vmlogin, s, 2) exited non-zero on 'SIP/17865221569-b6b03f60' in macro 'vmcenter' == Spawn extension (macro-vmlogin, s, 2) exited non-zero on 'SIP/17865221569-b6b03f60' Really destroying SIP dialog '[EMAIL PROTECTED]' Method: ACK On Mon, Oct 6, 2008 at 8:26 PM, Atis Lezdins <[EMAIL PROTECTED]> wrote: > On Tue, Oct 7, 2008 at 2:22 AM, Andrew Joakimsen <[EMAIL PROTECTED]> wrote: >> The odd thing is on this particular phone it only happens when you >> call voicemail. >> >> It is certainly a bug in Asterisk, not the UA. Asterisk is trying to >> send to 192.168.1.x which obviously is not possible. Something in the >> NAT support is not working right. > > Hi, > > You should get SIP traces to see why Asterisk is trying to reply to > 192.168.1.x. > > To do this, enter "sip set debug on" in asterisk CLI, and post us a > log of call reaching voicemail and disconnecting. > > Regards, > Atis > >> >> On Mon, Oct 6, 2008 at 3:06 PM, SIP <[EMAIL PROTECTED]> wrote: >>> This message is usually caused by Asterisk not receiving an ACK after >>> about 30 seconds of attempts. There are countless misconfigured UAs and >>> proxies out there that don't handle ACK well, so it would be nice to be >>> able to turn this 'feature' off. What's annoying is that the explanation >>> has always been "If we can't get an ACK, we can't send any RTP data." >>> This is patently false, as the RTP will often work fine even if ACK >>> handling is misconfigured (we see it all the time). >>> >>> But alas. As far as I can tell, there's no way to disable this check. I >>> suppose I could code around it, but not being the world's most >>> proficient C coder, I'm always afraid I'll break something else. ;) >>> >>> N. >>> >>> >>> Andrew Joakimsen wrote: >>>> 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 >>>> [EMAIL PROTECTED] for seqno 2 (Critical >>>> Response) >>>> [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1972 retrans_pkt: Hanging >>>> up call [EMAIL PROTECTED] - 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:[EMAIL PROTECTED] >>>> >>>> >>>> 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 >>>> >>>> _______________________________________________ >>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>>> >>>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >>>> Register Now: http://www.astricon.net >>>> >>>> asterisk-users mailing list >>>> To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>> >>> >>> >>> _______________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >>> Register Now: http://www.astricon.net >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > Atis Lezdins, > VoIP Project Manager / Developer, > [EMAIL PROTECTED] > Skype: atis.lezdins > Cell Phone: +371 28806004 > Cell Phone: +1 800 7300689 > Work phone: +1 800 7502835 > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users