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<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2>Hi All,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2> I've encountered the following very weird behaviour:
</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2>1. I have an Asterisk box located on the net, which is connected via SIP
to two endpoints.<BR> First endpoint is a SIPUA SPA-841 and
the other is a VERAZ softswitch.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2>2. When tyring to run a call from the Sipua to the VERAZ, it appears that
Asterisk tries<BR> to dial out no problem, but the following
WARNING is recieved on asterisk:</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2> May 3 14:55:35 WARNING[2136]: chan_sip.c:2934
process_sdp: Unknown SDP media type in offer: image 58232 udptl
t38<BR></FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2>3. Once this message is received, progress tones are no longer heard and
call disconnects<BR> on a one-way voice after 10
seconds.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2> I've ran a log debug on the call, and it looks like
this:</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2>May 3 14:55:32 DEBUG[2136]: chan_sip.c:5914 check_user_full:
Setting NAT on RTP to 0<BR>May 3 14:55:33 DEBUG[2136]: chan_sip.c:1014
__sip_ack: Stopping retransmission on <A
href="mailto:'c603084-839e381d@62.90.49.88'">'c603084-839e381d@62.90.49.88'</A>
of Response 101: Found<BR>May 3 14:55:33 DEBUG[2136]: chan_sip.c:5914
check_user_full: Setting NAT on RTP to 0<BR>May 3 14:55:33 DEBUG[2136]:
chan_sip.c:9113 handle_request: Ignoring too old SIP packet packet 101
(expecting >= 102)<BR>May 3 14:55:33 DEBUG[2136]: chan_sip.c:1014
__sip_ack: Stopping retransmission on <A
href="mailto:'c603084-839e381d@62.90.49.88'">'c603084-839e381d@62.90.49.88'</A>
of Response 102: Found<BR>May 3 14:55:33 DEBUG[2136]: chan_sip.c:5914
check_user_full: Setting NAT on RTP to 0<BR>May 3 14:55:33 DEBUG[2136]:
chan_sip.c:8600 handle_request_invite: Check for res for nirs<BR>May 3
14:55:33 DEBUG[2136]: chan_sip.c:5106 build_route: build_route: Contact hop:
nirs <sip:nirs@62.90.49.88:5060><BR> -- Executing
Dial("SIP/nirs-e218", "<A
href="mailto:SIP/902123400321@bveraz1|30">SIP/902123400321@bveraz1|30</A>") in
new stack<BR>May 3 14:55:33 DEBUG[2181]: chan_sip.c:1479 create_addr:
Setting NAT on RTP to 0<BR>May 3 14:55:33 DEBUG[2181]: chan_sip.c:1643
sip_call: Outgoing Call for 902123400321<BR> -- Called <A
href="mailto:902123400321@bveraz1">902123400321@bveraz1</A><BR>May 3
14:55:33 DEBUG[2136]: chan_sip.c:1060 __sip_semi_ack: (Provisional) Stopping
retransmission (but retaining packet) on <A
href="mailto:'4c1e49250ac249f22f48199407914f29@213.194.92.10'">'4c1e49250ac249f22f48199407914f29@213.194.92.10'</A>
Request 102: Found<BR>May 3 14:55:34 DEBUG[2136]: chan_sip.c:1060
__sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on
<A
href="mailto:'4c1e49250ac249f22f48199407914f29@213.194.92.10'">'4c1e49250ac249f22f48199407914f29@213.194.92.10'</A>
Request 102: Found<BR> -- SIP/bveraz1-b42b is
ringing<BR>May 3 14:55:35 DEBUG[2136]: chan_sip.c:1060 __sip_semi_ack:
(Provisional) Stopping retransmission (but retaining packet) on <A
href="mailto:'4c1e49250ac249f22f48199407914f29@213.194.92.10'">'4c1e49250ac249f22f48199407914f29@213.194.92.10'</A>
Request 102: Found<BR>May 3 14:55:35 WARNING[2136]: chan_sip.c:2934
process_sdp: Unknown SDP media type in offer: image 58232 udptl
t38<BR> -- SIP/bveraz1-b42b is making progress passing it to
SIP/nirs-e218<BR>May 3 14:55:53 DEBUG[2181]: chan_sip.c:1923 sip_hangup:
update_user_counter(902123400321) - decrement outUse counter<BR>May 3
14:55:53 DEBUG[2181]: app_dial.c:1345 dial_exec_full: Exiting with
DIALSTATUS=CANCEL.<BR> == Spawn extension (nirs, 902123400321, 1) exited
non-zero on 'SIP/nirs-e218'<BR>May 3 14:55:53 DEBUG[2181]: chan_sip.c:1926
sip_hangup: update_user_counter(nirs) - decrement inUse counter<BR>May 3
14:55:53 DEBUG[2136]: chan_sip.c:996 __sip_ack: Acked pending invite
102<BR>May 3 14:55:53 DEBUG[2136]: chan_sip.c:1014 __sip_ack: Stopping
retransmission on <A
href="mailto:'4c1e49250ac249f22f48199407914f29@213.194.92.10'">'4c1e49250ac249f22f48199407914f29@213.194.92.10'</A>
of Request 102: Found<BR>May 3 14:55:53 DEBUG[2136]: chan_sip.c:1014
__sip_ack: Stopping retransmission on <A
href="mailto:'4c1e49250ac249f22f48199407914f29@213.194.92.10'">'4c1e49250ac249f22f48199407914f29@213.194.92.10'</A>
of Request 102: Found<BR>May 3 14:55:53 DEBUG[2136]: chan_sip.c:1014
__sip_ack: Stopping retransmission on <A
href="mailto:'c603084-839e381d@62.90.49.88'">'c603084-839e381d@62.90.49.88'</A>
of Response 103: Found<BR></FONT></SPAN><SPAN class=178435409-03052005><FONT
face=Arial size=2></FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2>Now, SIP configuration looks like this:</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2>[general]<BR>;context=default
; Default context for incoming
calls<BR>realm=dimitel
; Realm for digest
authentication<BR>bindport=5060
; UDP Port to bind to (SIP standard port is
5060)<BR>bindaddr=0.0.0.0
; IP address to bind to (0.0.0.0 binds to
all)<BR>srvlookup=no
; Enable DNS SRV lookups on outbound calls<BR></FONT></SPAN><SPAN
class=178435409-03052005><FONT face=Arial
size=2>pedantic=yes
; Enable slow, pedantic checking for
Pingtel<BR>tos=lowdelay
;
lowdelay,throughput,reliability,mincost,none<BR>maxexpirey=3600
; Max length of incoming registration we
allow<BR>defaultexpirey=120
; Default length of incoming/outoing
registration<BR>checkmwi=10
; Default time between mailbox checks for peers<BR></FONT></SPAN><SPAN
class=178435409-03052005><FONT face=Arial
size=2>disallow=all
; First disallow all
codecs<BR>allow=ulaw
; Allow codecs in order of
preference<BR>allow=g729<BR>musicclass=default
; Sets the default music on hold class for all SIP
calls<BR>relaxdtmf=yes
; Relax dtmf
handling<BR>rtptimeout=60
; Terminate call if 60 seconds of no RTP
activity<BR>rtpholdtimeout=300
; Terminate call if 300 seconds of no RTP activity<BR>trustrpid =
no
; If Remote-Party-ID should be
trusted<BR>progressinband=never
; If we should generate in-band ringing always<BR>useragent=DimiTrex
iPBX ; Allows you to change the
user agent string<BR>dtmfmode =
info ; Set default dtmfmode for
sending DTMF. Default: rfc2833<BR></FONT></SPAN><SPAN
class=178435409-03052005><FONT face=Arial size=2>compactheaders
=no ; send compact sip
headers.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2>nat=no
; Global NAT settings (Affects all peers and
users)<BR>canreinvite=no</FONT></SPAN></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2>[nirs]<BR>type=friend<BR>host=dynamic<BR>nat=no<BR>canreinvinte=no<BR>username=nirs<BR>secret=nirs<BR>context=nirs</FONT></SPAN></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2>[bveraz1]<BR>type=friend<BR>host=62.244.xx.xx</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2>nat=no<BR>canreinvite=no<BR>disallow=all<BR>allow=g729</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2>And just for knowladge, I do have the g729 licenses installed on the
box.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2>Any thoughts on the issue would be highly
appreciated.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2>Regards,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=178435409-03052005><FONT face=Arial
size=2> Nir Simionovich</FONT></SPAN><SPAN
class=178435409-03052005></DIV></SPAN></BODY></HTML>