Chris Cooper 325
2011-Jan-28 18:43 UTC
[asterisk-users] asterisk-users Digest, Vol 78, Issue 66
It may have gone to sleep.
Chris Cooper
Systems/Network Administrator
EFC International
1940 Craigshire Blvd
St. Louis, MO 63146
US
Phone - 314-439-4325
Fax - 314-439-4443
Mobile - 314-402-8912
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Today's Topics:
1. Re: RTP keepalive doesn't work (Ryan Tucker)
2. Re: RTP keepalive doesn't work (Kevin P. Fleming)
3. Re: Caching CALLERID(dnid) (Olivier)
4. Re: Disabling Music On Hold (Urs Buob)
5. Re: chan_sip bug? (Asterisk 1.4) (Jian Gao)
6. How to disable srtp in asterisk 1.8.2.3? (Miguel Baptista)
7. Asterisk 1.8.2 - TLS, user certificate (Gilles ??)
----------------------------------------------------------------------
Message: 1
Date: Sat, 29 Jan 2011 01:24:18 +1000
From: Ryan Tucker <Ryan.Tucker at rgtech.com.au>
Subject: Re: [asterisk-users] RTP keepalive doesn't work
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <A2AF66A6-8790-4B41-9439-E4EEE1BA00DA at rgtech.com.au>
Content-Type: text/plain; charset="us-ascii"
Thanks for the info, I guess I would expect asterisk to send 'silence'
(in blank RTP form or something) if silence suppression is disabled. Just as I
would expect any end point to send 'silence' if it was muted when
silence suppression was disabled. It seems that RTP keepalives would serve this
purpose, however this doesn't seem to be available either... Should I file a
bug report re rtpkeepalive?
Sent from my iPhone
On 29/01/2011, at 12:55 AM, "Kevin P. Fleming" <kpfleming at
digium.com> wrote:
> On 01/27/2011 10:52 PM, Ryan Tucker wrote:
>> So, I've done some more testing and got some more info.
>>
>> I have one endpoint that does silence suppression and one that
doesn't. When the silence suppressing endpoint stops sending RTP, asterisk
stops sending RTP to the other endpoint. I have disabled directmedia and
directrtpsetup and it made no difference. I have even forced one endpoint to use
GSM and the other to use ULAW (forcing asterisk to re encode everything) and
asterisk STILL stops sending RTP when the endpoint does...
>
> Asterisk doesn't have anything to send. What do you expect it to send
> when it's not receiving anything? I see that we have an rtpkeepalive
> configuration option, but I don't see that any code actually causes
> keepalive packets to be sent anywhere... it did when it was first added,
> but somehow that code has been lost.
>
> This certainly warrants some investigation to find out when it was
> removed and why, because the configuration option should have been
> removed if the keepalive support was removed on purpose.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kfleming at digium.com
> Check us out at www.digium.com & www.asterisk.org
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> http://www.asterisk.org/hello
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> http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------
Message: 2
Date: Fri, 28 Jan 2011 09:32:56 -0600
From: "Kevin P. Fleming" <kpfleming at digium.com>
Subject: Re: [asterisk-users] RTP keepalive doesn't work
To: asterisk-users at lists.digium.com
Message-ID: <4D42E1A8.5040900 at digium.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
On 01/28/2011 09:24 AM, Ryan Tucker wrote:> Thanks for the info, I guess I would expect asterisk to send
'silence' (in blank RTP form or something) if silence suppression is
disabled. Just as I would expect any end point to send 'silence' if it
was muted when silence suppression was disabled. It seems that RTP keepalives
would serve this purpose, however this doesn't seem to be available
either... Should I file a bug report re rtpkeepalive?
No need... I'm already trying to track down when the code was removed,
and for what reason. Once that is done I'll enter an issue to get it
addressed.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfleming at digium.com
Check us out at www.digium.com & www.asterisk.org
------------------------------
Message: 3
Date: Fri, 28 Jan 2011 17:42:20 +0100
From: Olivier <oza_4h07 at yahoo.fr>
Subject: Re: [asterisk-users] Caching CALLERID(dnid)
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<AANLkTin-=bFYbBjAFyK2VcfZhcWCpyjpFZXJD20+Z4QS at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
2011/1/26 Arjan Kroon | Mobillion <Arjan.Kroon at mobillion.nl>
>
> Now we see that the CALLERID(dnid) is still '655871460'
>
> How do you exactly see that CALLERID(dnid) is still '655871460' ?
Are you reading it from the "called party side" or from from the
"calling
party side" ?
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Message: 4
Date: Fri, 28 Jan 2011 18:02:31 +0100
From: Urs Buob <Urs.Buob at ascom.CH>
Subject: Re: [asterisk-users] Disabling Music On Hold
To: asterisk-users at lists.digium.com
Message-ID:
<OFF4184A08.48C9E0D1-ONC1257826.005CF123-C1257826.005D9DA7 at
ln.ascom.ch>
Content-Type: text/plain; charset="us-ascii"
> On 11-01-28 07:37 AM, Urs Buob wrote:
> > modules.conf
> > ----------------------------------------------------------
> > [modules]
> > autoload=yes
> > ; res_phoneprov requires func_strings.so to be loaded:
> > preload => func_strings.so
> > noload => pbx_gtkconsole.so
> > noload => res_musiconhold.so
> >
> This is the correct method. But you are saying even if you stop and
> start Asterisk res_musiconhold.so is still loads?
>
> I just tested with the latest 1.6.2 branch with the same settings, MOH
> was not loaded.
Well, I did not say that MOH get's loaded. I just say that asterisk is
still trying to play MOH and does NOT inform the remote side of the hold
status.
Actually the error message that the CLI shows when I put the call on hold
also indicates that MOH is not loaded.
-- Music class default requested but no musiconhold loaded.
So, the problem is not that MOH is loaded, but that asterisk still tries
to invoke MOH (triggering the error message) and that there is no
re-invite to the remote SIP user indicating that the call is on hold. My
main goal is to have a clean hold functionality with re-invites that
asterisk sends out. (RTP stream goes via asterisk and not directly between
the SIP clients)
regards
Urs
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Message: 5
Date: Fri, 28 Jan 2011 09:20:22 -0800
From: Jian Gao <jian.gao at sjgeophysics.com>
Subject: Re: [asterisk-users] chan_sip bug? (Asterisk 1.4)
To: asterisk-users at lists.digium.com
Message-ID: <4D42FAD6.9030702 at sjgeophysics.com>
Content-Type: text/plain; charset="us-ascii";
Format="flowed"
Chad, You are right. tcpdump shows Asterisk sees 777 when the packet
arrived.
It's truned out my router somehow modified the packet! I am using a Asus
RT-N16 router with TomatoUSB firmware. There is a setting "SIP
Helper".
I disabled this "feature" on the router then everything back to
normal.
There is one thing still puzzle me: It seems enable or disable this
"feature" doesn't effect other SIP thunks. It could be the Sippy
server
use two different IP. The INVITE come from 208.65.xxx.xxx, but in its
packet it try to use 74.205.216.77 as contact address. Is my guess
correct? Why it does this?
*Jian *
On 11-01-27 04:31 PM, Chad Wallace wrote:> On Thu, 27 Jan 2011 14:52:06 -0800
> Jian Gao<jian.gao at sjgeophysics.com> wrote:
>
>> Today I upgraded my Asterisk to the new 1.4.39.1. One of sip trunk
>> stop working after the upgrade. Here is the sip debug:
>>
---------------------------------------------------------------------------
>> <--- SIP read from 208.65.xxx.xxx:5060 --->
> That packet is coming from the other end (Sippy). The problem is
> probably there. However, it could be that the networking routines in
> Asterisk have added a 7 at the end. You could compare a tcpdump of
> that packet to what Asterisk sees. If the tcpdump shows .777 then the
> problem is in Sippy. If it shows .77 then the problem is in Asterisk.
>
>
>> INVITE sip:1778xxxxxxx at 10.11.22.77:5060 SIP/2.0
>> Via: SIP/2.0/UDP
>>
208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-d9175178645e9146-1---d8754z-;rport
>> Via: SIP/2.0/UDP
>> 208.65.xxx.xxx:5061;branch=z9hG4bK-uhhmj2ir4ew6cn4p;rport=5061
>> Max-Forwards: 69
>> Record-Route:<sip:208.65.xxx.xxx;lr>
>> Contact: "Anonymous"<sip:208.65.xxx.xxx:5061>
>> To:<sip:1778xxxxxxx at 208.65.xxx.xxx:5060>
>> From:<sip:604xxxxxxx at
208.65.xxx.xxx:5060>;tag=ixpa27sbhn3inu5x.o
>> Call-ID: 550D37B3 at 208.72.xxx.xxx~o
>> CSeq: 819 INVITE
>> Expires: 300
>> Content-Disposition: session
>> Content-Type: application/sdp
>> User-Agent: Sippy
>> cisco-GUID: 2851810672-711266784-2763915291-559912524
>> h323-conf-id: 2851810672-711266784-2763915291-559912524
>> Content-Length: 109
>>
>> v=0
>> o=Sippy 223452192 0 IN IP4 74.205.216.77
>> s=-
>> t=0 0
>> m=audio 33830 RTP/AVP 0
>> c=IN IP4 74.205.216.777
>>
>> <------------->
>> --- (17 headers 6 lines) ---
>> Sending to 208.65.xxx.xxx : 5060 (NAT)
>> Using INVITE request as basis request - 550D37B3 at 208.72.xxx.xxx~o
>> Found peer 'FreePhoneLine'
>> Found RTP audio format 0
>> [2011-01-27 14:35:18] WARNING[2911]: chan_sip.c:5948 process_sdp_c:
>> Unable to lookup RTP Audio host in c= line, 'IN IP4
74.205.216.777'
>> [2011-01-27 14:35:18] WARNING[2911]: chan_sip.c:5741 process_sdp:
>> Insufficient information in SDP (c=)...
>>
-----------------------------------------------------------------------------------------------------------
>>
>>
>>
>>
>>
>> It seems in the SIP INVITE, the IP 74.205.216.77 somehow changed to
>> 74.205.216.777.
>> I am not sure this is a bug of Asterisk or not.
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
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Message: 6
Date: Fri, 28 Jan 2011 18:22:17 +0100
From: Miguel Baptista <miguel.baptista at uninett.no>
Subject: [asterisk-users] How to disable srtp in asterisk 1.8.2.3?
To: asterisk-users at lists.digium.com
Message-ID: <4D42FB49.9050503 at uninett.no>
Content-Type: text/plain; charset="iso-8859-1"
Hi all,
I upgraded one of our servers running asterisk 1.6.X to 1.8.2.3. I
compiled it with SRTP support.
Everything seems to work OK but I am having a weird issue. I cannot
disable SRTP. I tried the /encryption=no/ in /sip.conf /and the
/_SIPSRTP_CRYPTO=disable/ on my dailplan and it keeps trying to use the
SRTP.
Well, right now I have to have/ noload=res_srtp.so/ on my /modules.conf
/otherwise I cannot place SIP calls (cause other ends don't support it)
Regards,
Miguel Baptista
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Message: 7
Date: Sat, 29 Jan 2011 01:42:03 +0800
From: Gilles ?? <gvigner at gmail.com>
Subject: [asterisk-users] Asterisk 1.8.2 - TLS, user certificate
To: asterisk-users at lists.digium.com
Message-ID:
<AANLkTim4BopLXmQ=WXqVv_xYT+WtT4a1u31Kzh918QAu at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi guys,
In Malcolm Davenport Secure Calling
Tutorial<https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial#>,
a user certificate is generated and given to the client, but TLS works fine
without this certificate. So why should I use it and how to be sure it's
used ?
When trying to use it, I cannot see any client certificate request from the
server in Wireshark. Should it be set somewhere in the conf ? I couldn't
find anything in sip.conf or in Asterisk 1.8
doc<http://ofps.oreilly.com/titles/9780596517342/ch08.html#Voicemail_id272814)>about
it.
Thanks for your help,
Gilles
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