[Asterisk-Users] sip phone to sip phone errors

[Erick Perez]

Hi, the following logs are being generated while i test sip-to-sip
windows software phones.



Dec  7 17:05:16 WARNING[-159503440]: chan_sip.c:683 retrans_pkt:
Maximum retries exceeded on call
40dedd1535853f17250b4d0854e35c17 at 200.75.243.237 for seqno 102
(Critical Request)
  == No one is available to answer at this time
Dec  7 17:05:22 WARNING[-159503440]: chan_sip.c:683 retrans_pkt:
Maximum retries exceeded on call
40dedd1535853f17250b4d0854e35c17 at 200.75.243.237 for seqno 102
(Non-critical Request)
Dec  7 17:05:26 WARNING[-176354384]: pbx.c:1933 ast_pbx_run: Timeout,
but no rule 't' in context 'sip'
    -- Executing Dial("SIP/erick2-db3b", "SIP/erick1") in new stack
    -- Called erick1
Dec  7 17:05:56 WARNING[-159503440]: chan_sip.c:683 retrans_pkt:
Maximum retries exceeded on call
5184f4e66bc98ae361f5f5f512858a9f at 200.75.243.237 for seqno 102
(Critical Request)
  == No one is available to answer at this time
Dec  7 17:06:02 WARNING[-159503440]: chan_sip.c:683 retrans_pkt:
Maximum retries exceeded on call
5184f4e66bc98ae361f5f5f512858a9f at 200.75.243.237 for seqno 102
(Non-critical Request)
Dec  7 17:06:06 NOTICE[-176354384]: rtp.c:420 ast_rtp_read: RTP:
Received packet with bad UDP checksum
Dec  7 17:06:06 WARNING[-176354384]: pbx.c:1933 ast_pbx_run: Timeout,
but no rule 't' in context 'sip'
    -- Executing Dial("SIP/erick1-4afc", "SIP/erick2") in new stack
    -- Called erick2
    -- SIP/erick2-f752 is ringing
    -- SIP/erick2-f752 answered SIP/erick1-4afc
    -- Attempting native bridge of SIP/erick1-4afc and SIP/erick2-f752
Dec  7 17:08:13 WARNING[-159503440]: chan_sip.c:683 retrans_pkt:
Maximum retries exceeded on call
ECC33A73-9D42-40F4-BCBA-E1559ECC4B30 at 10.120.1.58 for seqno 53352
(Non-critical Response)
  == Spawn extension (sip, 2000, 1) exited non-zero on 'SIP/erick1-4afc'
    -- Executing Dial("SIP/erick2-efd9", "SIP/erick2") in new stack
    -- Called erick2
    -- SIP/erick2-9cf9 is ringing
    -- SIP/erick2-9cf9 answered SIP/erick2-efd9
    -- Attempting native bridge of SIP/erick2-efd9 and SIP/erick2-9cf9
    -- Started music on hold, class 'default', on SIP/erick2-9cf9
    -- Stopped music on hold on SIP/erick2-9cf9
  == Spawn extension (sip, 2000, 1) exited non-zero on 'SIP/erick2-efd9'
    -- Executing Dial("SIP/erick2-2685", "SIP/erick2") in new stack
    -- Called erick2
    -- SIP/erick2-eeae is ringing

they always drop the call.
suggestions?

thanks,