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 - -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of [email protected] Sent: Friday, January 28, 2011 12:00 PM To: [email protected] Subject: asterisk-users Digest, Vol 78, Issue 66 Send asterisk-users mailing list submissions to [email protected] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [email protected] You can reach the person managing the list at [email protected] When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-users digest..." 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 <[email protected]> Subject: Re: [asterisk-users] RTP keepalive doesn't work To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[email protected]> 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" <[email protected]> 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: [email protected] > Check us out at www.digium.com & www.asterisk.org > > -- > _____________________________________________________________________ > -- 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 > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ------------------------------ Message: 2 Date: Fri, 28 Jan 2011 09:32:56 -0600 From: "Kevin P. Fleming" <[email protected]> Subject: Re: [asterisk-users] RTP keepalive doesn't work To: [email protected] Message-ID: <[email protected]> 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: [email protected] Check us out at www.digium.com & www.asterisk.org ------------------------------ Message: 3 Date: Fri, 28 Jan 2011 17:42:20 +0100 From: Olivier <[email protected]> Subject: Re: [asterisk-users] Caching CALLERID(dnid) To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[email protected]> Content-Type: text/plain; charset="iso-8859-1" 2011/1/26 Arjan Kroon | Mobillion <[email protected]> > > 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" ? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110128/234a2dbb/attachment-0001.htm> ------------------------------ Message: 4 Date: Fri, 28 Jan 2011 18:02:31 +0100 From: Urs Buob <[email protected]> Subject: Re: [asterisk-users] Disabling Music On Hold To: [email protected] Message-ID: <off4184a08.48c9e0d1-onc1257826.005cf123-c1257826.005d9...@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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110128/a65fcfd0/attachment-0001.htm> ------------------------------ Message: 5 Date: Fri, 28 Jan 2011 09:20:22 -0800 From: Jian Gao <[email protected]> Subject: Re: [asterisk-users] chan_sip bug? (Asterisk 1.4) To: [email protected] Message-ID: <[email protected]> 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<[email protected]> 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:[email protected]: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:[email protected]:5060> >> From:<sip:[email protected]:5060>;tag=ixpa27sbhn3inu5x.o >> Call-ID: [email protected]~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 - [email protected]~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 > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110128/21ff9016/attachment-0001.htm> ------------------------------ Message: 6 Date: Fri, 28 Jan 2011 18:22:17 +0100 From: Miguel Baptista <[email protected]> Subject: [asterisk-users] How to disable srtp in asterisk 1.8.2.3? To: [email protected] Message-ID: <[email protected]> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110128/04e5bee5/attachment-0001.htm> ------------------------------ Message: 7 Date: Sat, 29 Jan 2011 01:42:03 +0800 From: Gilles ?? <[email protected]> Subject: [asterisk-users] Asterisk 1.8.2 - TLS, user certificate To: [email protected] Message-ID: <[email protected]> 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 -------------- next part -------------- An HTML attachment was scrubbed... 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If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _____________________________________________________________________ -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
