[Asterisk-Dev] Sip problem

Tom Dickenson voip at digitaldatabits.net
Thu Apr 7 21:53:07 MST 2005 

What does the following mean?  I get this message using a specific sip
service, calling any other service I do not get this message. I.E.
Stanaphone, Free World Dialup, FreeIPcall etc.

Apr  7 21:40:02 NOTICE[1330] chan_sip.c: '' is not a valid SIP contact
(missing sip:) trying to use anyway

-------------------------------------- Debug of
Call -------------------------------------
Apr  7 21:39:52 DEBUG[1330] acl.c: ##### Testing 192.168.1.112 with
192.168.0.0
Apr  7 21:39:52 DEBUG[1330] chan_sip.c: Setting NAT on RTP to 0
Apr  7 21:39:52 DEBUG[1330] chan_sip.c: Stopping retransmission on
'3396098948 at 192.168.1.112' of Response 1: Found
Apr  7 21:39:52 DEBUG[1330] chan_sip.c: Setting NAT on RTP to 0
Apr  7 21:39:52 DEBUG[1330] chan_sip.c: Check for res for cisco0
Apr  7 21:39:52 DEBUG[1330] chan_sip.c: build_route: Contact hop:
<sip:cisco0 at 192.168.1.112:5060;transport=udp>
Apr  7 21:39:52 VERBOSE[1344] logger.c: Asterisk Ready.
    -- Executing Macro("SIP/cisco0-e178", "speeddiall|15417260000") in new
stack
Apr  7 21:39:52 VERBOSE[1344] logger.c:     -- Executing
BackGround("SIP/cisco0-e178", "pls-wait-connect-call") in new stack
Apr  7 21:39:52 DEBUG[1344] channel.c: Scheduling timer at 160 sample
intervals
Apr  7 21:39:52 VERBOSE[1344] logger.c:     -- Playing
'pls-wait-connect-call' (language 'en')
Apr  7 21:39:52 DEBUG[1330] chan_sip.c: Stopping retransmission on
'3396098948 at 192.168.1.112' of Response 2: Found
Apr  7 21:39:54 DEBUG[1344] channel.c: Scheduling timer at 0 sample
intervals
Apr  7 21:39:54 DEBUG[1344] channel.c: Scheduling timer at 0 sample
intervals
Apr  7 21:39:54 VERBOSE[1344] logger.c:     -- Executing
Dial("SIP/cisco0-e178","SIP/15417260000 at mutualphone|50|r") in new stack
Apr  7 21:39:54 DEBUG[1344] chan_sip.c: Setting NAT on RTP to 524288
Apr  7 21:39:54 DEBUG[1344] acl.c: ##### Testing 209.250.147.116 with
192.168.0.0
Apr  7 21:39:54 DEBUG[1344] chan_sip.c: Target address 209.250.147.116 is
not local, substituting externip
Apr  7 21:39:54 DEBUG[1344] chan_sip.c: Outgoing Call for 15417260000
Apr  7 21:39:54 VERBOSE[1344] logger.c:     -- Called
15417260000 at mutualphone
Apr  7 21:39:54 DEBUG[1344] channel.c: Driver for channel 'SIP/cisco0-e178'
does not support indication 3, emulating it
Apr  7 21:39:54 DEBUG[1344] channel.c: Scheduling timer at 160 sample
intervals
Apr  7 21:39:54 DEBUG[1344] channel.c: Generator got voice, switching to
phase locked mode
Apr  7 21:39:54 DEBUG[1344] channel.c: Scheduling timer at 0 sample
intervals
Apr  7 21:39:54 DEBUG[1330] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'21c928cf068b1bef7af155f054eca082 at 63.227.222.221' Request 102: Found
Apr  7 21:39:54 DEBUG[1330] chan_sip.c: Acked pending invite 102
Apr  7 21:39:54 DEBUG[1330] chan_sip.c: Stopping retransmission on
'21c928cf068b1bef7af155f054eca082 at 63.227.222.221' of Request 102: Found
Apr  7 21:39:54 DEBUG[1330] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'21c928cf068b1bef7af155f054eca082 at 63.227.222.221' Request 103: Found
Apr  7 21:39:57 DEBUG[1330] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on
'21c928cf068b1bef7af155f054eca082 at 63.227.222.221' Request 103: Found
Apr  7 21:39:57 VERBOSE[1344] logger.c:     -- SIP/mutualphone-d0bf is
ringing
Apr  7 21:39:57 DEBUG[1344] rtp.c: RTP NAT: Using address
209.250.147.122:18840
Apr  7 21:40:02 DEBUG[1330] chan_sip.c: Acked pending invite 103
Apr  7 21:40:02 DEBUG[1330] chan_sip.c: Stopping retransmission on
'21c928cf068b1bef7af155f054eca082 at 63.227.222.221' of Request 103: Found
Apr  7 21:40:02 DEBUG[1330] chan_sip.c: build_route: Record-Route hop:
<sip:209.250.147.116:5060;lr>
Apr  7 21:40:02 DEBUG[1330] chan_sip.c: build_route: Contact hop:
<sip:0015417260000 at 209.250.147.122;user=phone>
Apr  7 21:40:02 VERBOSE[1344] logger.c:     -- SIP/mutualphone-d0bf answered
SIP/cisco0-e178
Apr  7 21:40:02 DEBUG[1344] channel.c: Scheduling timer at 0 sample
intervals
Apr  7 21:40:02 VERBOSE[1344] logger.c:     -- Attempting native bridge of
SIP/cisco0-e178 and SIP/mutualphone-d0bf
Apr  7 21:40:02 DEBUG[1330] chan_sip.c: Acked pending invite 102
Apr  7 21:40:02 DEBUG[1330] chan_sip.c: Stopping retransmission on
'3396098948 at 192.168.1.112' of Request 102: Found
Apr  7 21:40:02 DEBUG[1330] chan_sip.c: build_route: Contact hop:
<sip:cisco0 at 192.168.1.112:5060;transport=udp>
Apr  7 21:40:02 DEBUG[1330] chan_sip.c: Acked pending invite 104
Apr  7 21:40:02 DEBUG[1330] chan_sip.c: Stopping retransmission on
'21c928cf068b1bef7af155f054eca082 at 63.227.222.221' of Request 104: Found
Apr  7 21:40:02 NOTICE[1330] chan_sip.c: '' is not a valid SIP contact
(missing sip:) trying to use anyway
Apr  7 21:40:02 DEBUG[1330] chan_sip.c: build_route: Retaining previous
route: <sip:209.250.147.116:5060;lr>
Apr  7 21:40:11 DEBUG[1344] channel.c: Returning from native bridge,
channels: SIP/cisco0-e178, SIP/mutualphone-d0bf
Apr  7 21:40:11 DEBUG[1344] chan_sip.c: update_user_counter(15417260000) -
decrement outUse counter
Apr  7 21:40:11 DEBUG[1344] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Apr  7 21:40:11 VERBOSE[1344] logger.c:   == Spawn extension
(macro-speeddiall, s, 2) exited non-zero on 'SIP/cisco0-e178' in macro
'speeddiall'
Apr  7 21:40:11 VERBOSE[1344] logger.c:   == Spawn extension (local, 323, 1)
exited non-zero on 'SIP/cisco0-e178'
Apr  7 21:40:11 DEBUG[1344] chan_sip.c: update_user_counter(cisco0) -
decrement outUse counter
Apr  7 21:40:11 DEBUG[1330] chan_sip.c: Acked pending invite 105
Apr  7 21:40:11 DEBUG[1330] chan_sip.c: Stopping retransmission on
'21c928cf068b1bef7af155f054eca082 at 63.227.222.221' of Request 105: Found
Apr  7 21:40:11 NOTICE[1330] chan_sip.c: '' is not a valid SIP contact
(missing sip:) trying to use anyway
Apr  7 21:40:11 DEBUG[1330] chan_sip.c: build_route: Retaining previous
route: <sip:209.250.147.116:5060;lr>
Apr  7 21:40:11 DEBUG[1330] chan_sip.c: Stopping retransmission on
'21c928cf068b1bef7af155f054eca082 at 63.227.222.221' of Request 106: Found


----- Original Message ----- 
From: "Tom Dickenson" <voip at digitaldatabits.net>
To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
Sent: Thursday, April 07, 2005 6:55 PM
Subject: [Asterisk-Dev] Sip problem


> Is there something wrong with my sip configuration or is this a bug? Can
> someone help me out here with this strange message... I DoNt UnDeRsTaNd...
>
> Apr 7 17:44:08 NOTICE[692]: chan_sip.c 4810 parse_ok_contact: '' is not a
> valid SIP contact (missing sip:) trying to use anyway
> Apr 7 17:50:02 NOTICE[692]: chan_sip.c 4810 parse_ok_contact: '' is not a
> valid SIP contact (missing sip:) trying to use anyway
>
>
> - Tom Dickenson
>
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