[Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
Sean Devoy
sdevoy at bizfocused.com
Thu Dec 20 20:31:35 MSK 2012
First, THANKS to all for ideas.
Second, I am at firmware 7.4.n Cisco has jumped to 7.5.3 I had no idea it
was so volatile. I will upgrade on Saturday.
Third, Anthony M asked for the sip trace in the file. I do ANYTHING he
wants!
Can someone spend 3 minutes and explain the relationships and functions of:
1. Sofia loglevel
2. Sofia tracelevel
3. Console loglevel
4. Sofia global siptrace (and sofia prfile XYZ siptrace on|off)
Thanks,
Sean
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, December 20, 2012 11:50 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
On Thu, Dec 20, 2012 at 8:36 AM, Steven Ayre <steveayre at gmail.com> wrote:
Why does it only happen occasionally (even from this same caller)?
That makes me wonder if there's a bug that rarely treats Block Anonymous
Call as Yes. Cisco has been known for buggy SIP implementations in the past.
See if there's a firmware upgrade available.
Otherwise it's also possible it's disliking something set in the
Allow/Supported/Allow-Events headers. I would take a siptrace of a working
call to the same user and compare the INVITE packets to see if there's a
difference.
You don't need the sofia stack debugging on by the way, which'll make your
logs much smaller.
Pro tip:
For SIP trace just use something like:
sofia global siptrace on
-OR-
sofia profile internal siptrace on
Don't do "sofia loglevel xxx" if you're just wanting a simple SIP trace.
-MC
-Steve
On 20 December 2012 15:39, Sean Devoy <sdevoy at bizfocused.com> wrote:
Thank you very much for your response Steven. I have checked the Block
Anonymous Call and it is set to NO.
But, at least I know it is the 504G's issue and I can dig deeper there. I
will certainly start with the firmware versions, etc.
Can you take a guess at something for me?
Why does it only happen occasionally (even from this same
caller)?
Is this the same SIP response to a phone on Do Not Disturb?
Thanks again.
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
Ayre
Sent: Thursday, December 20, 2012 9:50 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
The Cisco SPA504G doesn't like something in this request and gives the 406
Not Acceptable response (SERVICE_NOT_IMPLEMENTED is the Q.931 mapping of SIP
406).
INVITE sip:302 at 69.251.170.6:1085 SIP/2.0
Via: SIP/2.0/UDP 204.62.15.231;rport;branch=z9hG4bKaFrHeKgQ6F2Ue
Max-Forwards: 70
From: "anonymous" <sip:4104850152 at 204.62.15.231
<mailto:sip%3A4104850152 at 204.62.15.231> >;tag=aK7263m4r24Hj
To: <sip:302 at 69.251.170.6:1085>
Call-ID: 8308fa5b-c3ed-1230-899f-03777f384a29
CSeq: 37593727 INVITE
Contact: <sip:mod_sofia at 204.62.15.231:5060>
User-Agent:
FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120
712T101002Z
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: precondition, path, replaces
Allow-Events: talk, hold, conference, presence, dialog, line-seize,
call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 209
X-FS-Support: update_display,send_info
Remote-Party-ID: "anonymous" <sip:4104850152 at 204.62.15.231
<mailto:sip%3A4104850152 at 204.62.15.231>
>;party=calling;screen=yes;privacy=full
v=0
o=FreeSWITCH 1355837829 1355837830 IN IP4 204.62.15.231
s=FreeSWITCH
c=IN IP4 204.62.15.231
t=0 0
m=audio 21754 RTP/AVP 9 0 8 3 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
This site suggests that it could be blocking the call because the callerid
is "anonymous""
http://www.mysysadmintips.com/linux/servers/309-linksys-spa-voip-phone-rejec
ts-calls-from-skype-sip-trunk-on-asterisk
That page shows a setting to disable that, can see you if that helps?
You could also try setting effective_caller_id_name to send something other
than anonymous.
-Steve
On 19 December 2012 22:33, Sean Devoy <sdevoy at bizfocused.com> wrote:
Sorry, server won't serve .log files!
http://www.bizfocused.com/service_not_found.txt
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, December 19, 2012 4:50 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
No the url is 404 so I can't see the file.
On Wed, Dec 19, 2012 at 3:43 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
Whoa. 404 a.k.a device not found?
This could be as I suspected, router capacity exceeded or general NAT issue
with this router?
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, December 19, 2012 1:27 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
it gives 404 err
On Tue, Dec 18, 2012 at 3:37 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
OK, I got one the logs! I have trimmed the file down, but it is still over
5 MB and pastebin just chokes on it.
It is on my server: http://www.bizfocused.com/service_not_found.log
Thanks for looking.
If you really need pcap, I am afraid I need detailed instructions on
installing/running it.
Sean
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 17, 2012 7:00 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
usually the 406 is done in an established call to refuse a codec change
during re-invite.
Its possible the other end thinks we want to change the codec when we do the
session-timer re-invite but I'm sure we don't but the sip trace can help
shed some light. You can run a pcap too at the same time so when we find
the bad call in the logs we can filter it out of the pcap too. To avoid it
getting too big you can just restart it every so often or use sippcapdump
and delete calls that are not affected.
On Mon, Dec 17, 2012 at 5:51 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
Waiting for another failure with siptrace REALLY on this time.
If the user has clicked DND on these cisco phones, could that cause these
messages?
Sean
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, December 17, 2012 12:32 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
The siptrace is not in here? Did you enable it on the console and then
capture to the log file perhaps?
you should do "sofia tracelevel debug" too to route traces to the log file.
Its hard to say for sure with no sip trace but it seems like the far end is
rejecting the session timer re-invite causing the call to end.
On Mon, Dec 17, 2012 at 11:05 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:
http://pastebin.freeswitch.org/20342
--
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Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
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_________________________________________________________________________
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Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_minessale at hotmail.com
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_________________________________________________________________________
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