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.
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:
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.
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 '[EMAIL PROTECTED]' 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 '[EMAIL PROTECTED]' 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:[EMAIL PROTECTED]:5060>
-- Executing Dial("SIP/nirs-e218", "SIP/[EMAIL PROTECTED]|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 [EMAIL PROTECTED]
May 3 14:55:33 DEBUG[2136]: chan_sip.c:1060 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found
May 3 14:55:34 DEBUG[2136]: chan_sip.c:1060 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' 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 '[EMAIL PROTECTED]' 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 '[EMAIL PROTECTED]' of Request 102: Found
May 3 14:55:53 DEBUG[2136]: chan_sip.c:1014 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found
May 3 14:55:53 DEBUG[2136]: chan_sip.c:1014 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 103: Found
May 3 14:55:33 DEBUG[2136]: chan_sip.c:1014 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' 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 '[EMAIL PROTECTED]' 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:[EMAIL PROTECTED]:5060>
-- Executing Dial("SIP/nirs-e218", "SIP/[EMAIL PROTECTED]|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 [EMAIL PROTECTED]
May 3 14:55:33 DEBUG[2136]: chan_sip.c:1060 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found
May 3 14:55:34 DEBUG[2136]: chan_sip.c:1060 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' 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 '[EMAIL PROTECTED]' 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 '[EMAIL PROTECTED]' of Request 102: Found
May 3 14:55:53 DEBUG[2136]: chan_sip.c:1014 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found
May 3 14:55:53 DEBUG[2136]: chan_sip.c:1014 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' 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.
;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
canreinvite=no
[nirs]
type=friend
host=dynamic
nat=no
canreinvinte=no
username=nirs
secret=nirs
context=nirs
type=friend
host=dynamic
nat=no
canreinvinte=no
username=nirs
secret=nirs
context=nirs
[bveraz1]
type=friend
host=62.244.xx.xx
type=friend
host=62.244.xx.xx
nat=no
canreinvite=no
disallow=all
allow=g729
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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