Home user - calls to cell phone drop

asterisk Version 11.2
FreePBX 2.11.10.11
Service is a 23 Channel PRI
Problem User is on a 20M down/2M up connection at home

Our PBX supports about 100 users and works well, except for one remote user.
60 Users local, about 30 users remote office over VPN and 10 or so remote individual users

Server is in MA, user is in CA. They connect remotely over SIP to the outside network and firewall rules allow his IP.

The Good

  1. Anyone can call him from a land line or a cell phone. no problem
  2. He can call anyone on the PBX , no problem
  3. he Can call another outside land line (he called my office line), no problem

The Bad

  1. if he calls a cell phone from his IP Phone, they answer and about 5-10 seconds it disconnects . He replicated this with me. 3-5 seconds after a cell phone answers it hangs up. During that time, the log file shows this.

[2014-05-15 11:49:38] WARNING[2268] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 2 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 7551ms with no response

—Because all other phones work, and it only happens with him outbound to a cell phone, my theory is that it has to do with CA->MA->Cell Network = too much delay?

My current thought is to put him behind a hardware VPN , but I am looking for other insight.

–LOGS

(**note number and IP address changed below for privacy)
2014-05-15 11:49:22] VERBOSE[28421][C-00002363] pbx.c: – Executing [[email protected]:22] Dial(“SIP/498-000002da”, “DAHDI/r0/15555555555,300,Tt”) in new stack
[2014-05-15 11:49:22] VERBOSE[28421][C-00002363] sig_pri.c: – Requested transfer capability: 0x00 - SPEECH
[2014-05-15 11:49:22] VERBOSE[28421][C-00002363] app_dial.c: – Called DAHDI/r0/15555555555
[2014-05-15 11:49:22] VERBOSE[28421][C-00002363] app_dial.c: – DAHDI/i1/15555555555-1c3 is proceeding passing it to SIP/498-000002da
[2014-05-15 11:49:23] VERBOSE[28421][C-00002363] app_dial.c: – DAHDI/i1/15555555555-1c3 is ringing
[2014-05-15 11:49:23] VERBOSE[28421][C-00002363] app_dial.c: – DAHDI/i1/15555555555-1c3 is making progress passing it to SIP/498-000002da
[2014-05-15 11:49:31] VERBOSE[28421][C-00002363] app_dial.c: – DAHDI/i1/15555555555-1c3 answered SIP/498-000002da
[2014-05-15 11:49:38] WARNING[2268] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 2 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 7551ms with no response
[2014-05-15 11:49:38] WARNING[2268] chan_sip.c: Hanging up call [email protected] - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[2014-05-15 11:49:38] VERBOSE[28421][C-00002363] pbx.c: – Executing [[email protected]:1] Macro(“SIP/498-000002da”, “hangupcall,”) in new stack
[2014-05-15 11:49:38] VERBOSE[28421][C-00002363] pbx.c: – Executing [[email protected]:1] GotoIf(“SIP/498-000002da”, “1?theend”) in new stack
[2014-05-15 11:49:38] VERBOSE[28421][C-00002363] pbx.c: – Goto (macro-hangupcall,s,3)
[2014-05-15 11:49:38] VERBOSE[28421][C-00002363] pbx.c: – Executing [[email protected]:3] ExecIf(“SIP/498-000002da”, “0?Set(CDR(recordingfile)=)”) in new stack
[2014-05-15 11:49:38] VERBOSE[28421][C-00002363] pbx.c: – Executing [[email protected]:4] Hangup(“SIP/498-000002da”, “”) in new stack
[2014-05-15 11:49:38] VERBOSE[28421][C-00002363] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/498-000002da’ in macro ‘hangupcall’
[2014-05-15 11:49:38] VERBOSE[28421][C-00002363] pbx.c: == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on ‘SIP/498-000002da’
[2014-05-15 11:49:38] VERBOSE[28421][C-00002363] chan_dahdi.c: – Hungup ‘DAHDI/i1/15555555555-1c3’
[2014-05-15 11:49:38] VERBOSE[28421][C-00002363] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on ‘SIP/498-000002da’ in macro ‘dialout-trunk’
[2014-05-15 11:49:38] VERBOSE[28421][C-00002363] pbx.c: == Spawn extension (from-internal, 16173782662, 5) exited non-zero on ‘SIP/498-000002da’
[2014-05-15 11:49:55] VERBOSE[2268][C-00002364] netsock2.c: == Using SIP RTP TOS bits 184

Nothing to do with delay, your asterisk box is not able to maintain the rtp stream through the internet, check your timeouts and keep-alive rtp settings both on asterisk and any routers involved, make sure you are not trying to reinvite your media path, cutting your Asterisk box out of the picture, which won’t work over dahdi.

thanks dicko… I will check these settings.