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der output (4 bits in the first byte) ???=0A=A0=0ASo here is a scenario I w= ould like some feedback on:=0A=A0=0AClient A makes a narrowband Speex VOIP = call to Client B.=0A=A0=0AIn the SIP negotiation there is no information ex= changed besides the fact that it is a narrowband speex call (speex/8000)=0A= =A0=0AClient A operates at a bitrate of 11kbit/s (28 bytes per payload)=0A= =A0=0AClient B

it was mentioned that echo cancellation will not work if the playback and capture is done on 2 different soundcard. I would like to know whether this is true. I trying to use Speex Echo Cancellation on capture audio from a webcam mic with the playback through my onboard audio. Based on what is mentioned in the troubleshooting guide, does this mean that I would never be able to get the Echo

/var/lib/asterisk/sounds/soundfile.alaw /var/lib/asterisk/sounds/soundfile.wav to go from alaw to mp3, first convert to wav, then use lame <options> /var/lib/asterisk/sounds/soundfile.wav /var/lib/asterisk/sounds/soundfile.mp3 sox looks like it can ogg/vorbis, but mine doesn't list mp3. You might fetch the source for sox and see if it can do mp3; lame is probably just as easy to obtain

--timeout=TIMEOUT This option allows you to set a maximum IO timeout in seconds. If no data is transferred for the specified time then rsync will exit. The default is 0, which means no timeout. Could anyone please tell me why IO timeout occures..in other words, why data could not be transferred in 12 hours? BTW, I have not set IOTIMEOUT from rsync but still it has been existing.

faced this exact same problem a few times on more than one servers and it was 1) dialplan issue which was not hanging up the zap channels correctly 2) using more than 8 spans on a server. Asterisk can't handle more than 96 zap channels on T1s. FXO/FXS combinations can vary the number of spans but if you know what I mean by spans, in production don't use more than 6 spans. On 2010-03-17

crashing. So, as a first step to solving **that** problem, make sure asterisk is compiled with debug flags, dumps another core file, and then you do the "gdb asterisk <corefilename>", and get a stack trace. That should give us some idea of what happened. > > I have a fairly simple Followme sequence in place to see how it works > before I get into the complex scenarios.

supported, beside Idle, On call and Ringing ? Can we expect this list to match DEVICE_STATE's one (UNKNOWN | NOT_INUSE | INUSE | BUSY | INVALID | UNAVAILABLE | RINGING | RINGINUSE | ONHOLD) > Might be worth seeing if other phones do the same. > > S > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by

"Why Siphon doesn't allow to receive a call when it doesn't run Apple doesn't accept (for the moment) an application runs in the background= . So, when Siphon doesn't run, the SIP server of your provider doesn't know your iPhone." --0015174c3c60a73ef5046656ca27 Content-Type: text/html; charset=windows-1252 Content-Transfer-Encoding: quoted-printable

connection will remain a TCP connection unless it is broken and restarted. Usually if I stop the client and wait for about 30 seconds to reconnect, there is a much greater chance that the MTU probes work fine, and in about 30 seconds MTU is fixed to 1416. Every time when the MTU probing fails, I see latency between 700 - 1000 ms with 32 byte pings over a LAN. Every time when the MTU probing does

have problems with outgoing calls. When I tried this, the same way you did, I could make calles externally but had no audio each way reguardless of what I tried to pass to the sip provider. Best bet is to use what your sip provider can use or find another provider that that can do g722. That's what I did when I wanted to use g726. my2cents On Tue, Jun 29, 2010 at 2:42 PM, Mindaugas Kezys

*is* on a network computer A belongs to, the 10.30.1.0/24 network. So it will do an ARP request (broadcast) to get the MAC address associated with the 10.30.1.130 IP. The local Tinc gateway will ultimately (I believe, Guus can speak on this with more authority than I can) perform the job of proxy ARP to get the traffic to the destination on the other side of the VPN. Regards, Donald On Thu,

[0004f2xxxxxx](poly650) defaultuser=0004f2xxxxxx callerid="Front Desk" <1600> mailbox=1600 *setvar=callidnum=1234561600* and from extensions.conf: [outgoing] ; Outbound unrestricted domestic calls exten => _1NXXXXXXXXX,1,Verbose(Outbound call from ${callidnum} to ${EXTEN} on ${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)}.) *exten =>

typing his number, though there is a 15 seconds timeout, and even if I type the number very fast it still may happen to me. *same => n,Read(mobileNumber,app/input-mobile,10,,2,15)* In the logs: When it fails: - - <SIP/ipbx-iwred-000002e> Playing 'app/input-mobile.slin' (language 'fr') - - User disconnected When it succeeds: - - <SIP/ipbx-iwred-000002e> Playing

be able to setup a SIP trunk. I've been able to successfully integrate a Cisco CallManager 7.x system with Asterisk using SIP trunking, so I imagine you should be able to do the same here. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com --0016e651f0a6bbe47b04a303939e Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: quoted-printable <div

Emscripten is an LLVM-to-JavaScript compiler. It takes LLVM bitcode - which can be generated from C/C++, using llvm-gcc or clang, or any other language that can be converted into LLVM - and compiles that into JavaScript, which can be run on the web (or anywhere else JavaScript can run). I was able to successfully build libogg, libvorbis and libvorbis examples using this tool and generate valid

<br> [memberconnector]<br> ;<br> exten =3D&gt; _XXX,1,Dial(SIP/${peerPrefix}$<u></u>{EXTEN},${TIMERINGQUEUE}= ,)<br> =A0 =A0 =A0 same =3D&gt; n,NoOp(DIALSTATUS=3D${<u></u>DIALSTATUS})<br> <br> As you can see, all status are empty,<div class=3D""><div><br> <br> -- <br>

built-in; This doesn=92t matter because the moderator would have to use meetmeadmin or the confbridge equivalent to control the other functions. The moderator would either need two phones or a phone and a web = interface. Let=92s say Yves=92 =93special conference=94 is 5555. The moderator = would start using this command Exten =3D> s,1,meetme(5555) The participants would do Exten =3D>

at least once a week I receive such an attack coming from a different ip. I will read the articles. Thanks again to everyone. Regards, Rodrigo Lang. 2010/6/29 Kenny Watson <kwatson at geniusgroupltd.com> > Hi, you can use fail2ban >

could be "hot". Is there any chance this would cause the card to fail after a while? It appears this site just had 4 port Digium card fail today. > Also, I am trying to cross connect with another Asterisk system with > > the normal LBO setting (i.e. span=1,1,0,esf,b8zs) but as of yet the > > systems aren't seeing each other at all. Could the side with the high >

r> <h2>Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010= ) </h2>With SIP 3.2.X firmware (available on the Polycom download site)=20 and Asterisk 1.6.1, Polycom phones now support a full featured BLF=20 showing statuses of Ringing, Inuse and Online and one touch directed=20 call pickup. <br>On the asterisk side all that needs to be done is to add a hint