[Asterisk-Users] Very weird behaviour of Asterisk and SIP

Tue May 3 04:08:02 MST 2005

Hi,
I had similar problem with today cvs. When I make call gs phones
eachother no voice both direction.
I downgraded to yesterday cvs.

On Tue, 2005-05-03 at 12:01 +0200, Nir Simionovich wrote:
> Hi All,
>  
>   I've encountered the following very weird behaviour: 
>  
> 1. I have an Asterisk box located on the net, which is connected via
> SIP to two endpoints.
>     First endpoint is a SIPUA SPA-841 and the other is a VERAZ
> softswitch.
> 2. When tyring to run a call from the Sipua to the VERAZ, it appears
> that Asterisk tries
>     to dial out no problem, but the following WARNING is recieved on
> asterisk:
>  
>     May  3 14:55:35 WARNING[2136]: chan_sip.c:2934 process_sdp:
> Unknown SDP media type in offer: image 58232 udptl t38
> 
> 3. Once this message is received, progress tones are no longer heard
> and call disconnects
>     on a one-way voice after 10 seconds.
>  
>   I've ran a log debug on the call, and it looks like this:
>  
> May  3 14:55:32 DEBUG[2136]: chan_sip.c:5914 check_user_full: Setting
> NAT on RTP to 0
> May  3 14:55:33 DEBUG[2136]: chan_sip.c:1014 __sip_ack: Stopping
> retransmission on 'c603084-839e381d at 62.90.49.88' of Response 101:
> Found
> May  3 14:55:33 DEBUG[2136]: chan_sip.c:5914 check_user_full: Setting
> NAT on RTP to 0
> May  3 14:55:33 DEBUG[2136]: chan_sip.c:9113 handle_request: Ignoring
> too old SIP packet packet 101 (expecting >= 102)
> May  3 14:55:33 DEBUG[2136]: chan_sip.c:1014 __sip_ack: Stopping
> retransmission on 'c603084-839e381d at 62.90.49.88' of Response 102:
> Found
> May  3 14:55:33 DEBUG[2136]: chan_sip.c:5914 check_user_full: Setting
> NAT on RTP to 0
> May  3 14:55:33 DEBUG[2136]: chan_sip.c:8600 handle_request_invite:
> Check for res for nirs
> May  3 14:55:33 DEBUG[2136]: chan_sip.c:5106 build_route: build_route:
> Contact hop: nirs <sip:nirs at 62.90.49.88:5060>
>     -- Executing Dial("SIP/nirs-e218", "SIP/902123400321 at bveraz1|30")
> in new stack
> May  3 14:55:33 DEBUG[2181]: chan_sip.c:1479 create_addr: Setting NAT
> on RTP to 0
> May  3 14:55:33 DEBUG[2181]: chan_sip.c:1643 sip_call: Outgoing Call
> for 902123400321
>     -- Called 902123400321 at bveraz1
> May  3 14:55:33 DEBUG[2136]: chan_sip.c:1060 __sip_semi_ack:
> (Provisional) Stopping retransmission (but retaining packet) on
> '4c1e49250ac249f22f48199407914f29 at 213.194.92.10' Request 102: Found
> May  3 14:55:34 DEBUG[2136]: chan_sip.c:1060 __sip_semi_ack:
> (Provisional) Stopping retransmission (but retaining packet) on
> '4c1e49250ac249f22f48199407914f29 at 213.194.92.10' Request 102: Found
>     -- SIP/bveraz1-b42b is ringing
> May  3 14:55:35 DEBUG[2136]: chan_sip.c:1060 __sip_semi_ack:
> (Provisional) Stopping retransmission (but retaining packet) on
> '4c1e49250ac249f22f48199407914f29 at 213.194.92.10' Request 102: Found
> May  3 14:55:35 WARNING[2136]: chan_sip.c:2934 process_sdp: Unknown
> SDP media type in offer: image 58232 udptl t38
>     -- SIP/bveraz1-b42b is making progress passing it to SIP/nirs-e218
> May  3 14:55:53 DEBUG[2181]: chan_sip.c:1923 sip_hangup:
> update_user_counter(902123400321) - decrement outUse counter
> May  3 14:55:53 DEBUG[2181]: app_dial.c:1345 dial_exec_full: Exiting
> with DIALSTATUS=CANCEL.
>   == Spawn extension (nirs, 902123400321, 1) exited non-zero on
> 'SIP/nirs-e218'
> May  3 14:55:53 DEBUG[2181]: chan_sip.c:1926 sip_hangup:
> update_user_counter(nirs) - decrement inUse counter
> May  3 14:55:53 DEBUG[2136]: chan_sip.c:996 __sip_ack: Acked pending
> invite 102
> May  3 14:55:53 DEBUG[2136]: chan_sip.c:1014 __sip_ack: Stopping
> retransmission on '4c1e49250ac249f22f48199407914f29 at 213.194.92.10' of
> Request 102: Found
> May  3 14:55:53 DEBUG[2136]: chan_sip.c:1014 __sip_ack: Stopping
> retransmission on '4c1e49250ac249f22f48199407914f29 at 213.194.92.10' of
> Request 102: Found
> May  3 14:55:53 DEBUG[2136]: chan_sip.c:1014 __sip_ack: Stopping
> retransmission on 'c603084-839e381d at 62.90.49.88' of Response 103:
> Found
> 
> Now, SIP configuration looks like this:
>  
> [general]
> ;context=default                        ; Default context for incoming
> calls
> realm=dimitel           ; Realm for digest authentication
> bindport=5060                   ; UDP Port to bind to (SIP standard
> port is 5060)
> bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds
> to all)
> srvlookup=no                    ; Enable DNS SRV lookups on outbound
> calls
> pedantic=yes                    ; Enable slow, pedantic checking for
> Pingtel
> tos=lowdelay                    ;
> lowdelay,throughput,reliability,mincost,none
> maxexpirey=3600         ; Max length of incoming registration we allow
> defaultexpirey=120              ; Default length of incoming/outoing
> registration
> checkmwi=10                     ; Default time between mailbox checks
> for peers
> disallow=all                    ; First disallow all codecs
> allow=ulaw                      ; Allow codecs in order of preference
> allow=g729
> musicclass=default              ; Sets the default music on hold class
> for all SIP calls
> relaxdtmf=yes                   ; Relax dtmf handling
> rtptimeout=60                   ; Terminate call if 60 seconds of no
> RTP activity
> rtpholdtimeout=300              ; Terminate call if 300 seconds of no
> RTP activity
> trustrpid = no                  ; If Remote-Party-ID should be trusted
> progressinband=never            ; If we should generate in-band
> ringing always
> useragent=DimiTrex iPBX         ; Allows you to change the user agent
> string
> dtmfmode = info         ; Set default dtmfmode for sending DTMF.
> Default: rfc2833
> compactheaders =no      ; send compact sip headers.
> nat=no                          ; Global NAT settings  (Affects all
> peers and users)
> canreinvite=no
>  
> [nirs]
> type=friend
> host=dynamic
> nat=no
> canreinvinte=no
> username=nirs
> secret=nirs
> context=nirs
>  
> [bveraz1]
> type=friend
> host=62.244.xx.xx
> nat=no
> canreinvite=no
> disallow=all
> allow=g729
>  
> And just for knowladge, I do have the g729 licenses installed on the
> box.
> Any thoughts on the issue would be highly appreciated.
>  
> Regards,
>   Nir Simionovich
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