Re: [asterisk-users] MagicJack and Skype call quality
On Sat, 12 Jul 2008 10:54:07 -0400, Julio Arruda wrote: >Jason Aarons (US) wrote: >> My understanding is Skype's secret is using the iLBC codec, which Cisco >> has also licensed for their 79X2 models as well. I travel and lot and >> in places where Yahoo Phone Out or MSN Phone or Cisco IP Communicator >> will fail the Skype client will work. The iLBC codec can really handle >> packet loss. > >AFAIK, not iLBC, but another GIPS codec. >ILBC is present in some grandstream phones from what I remember, not in >the Cisco 7912 as one example, not sure about other phones. >In a word with still many PSTN gateways out there that don't support it, >ILBC in the SIP UA side only can help that much :-).. I recall reading somewhere that in v3 they abandonned the GIPS codec in favor of something else. It's still included for backward compatability, but they have a newer prefered codec. Michael -- Michael Graves mgravesmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming call does not reach asterisk.
On Sun, 13 Jul 2008, Chris Rowson wrote: > Hi, this is my first post to the list, but I have tried to search > elsewhere for a solution, and have had a read of 'Asterisk - The > Future of Telephony'. So you could say that I have at least tried to > RTFM as it were! > > I've configured a couple of Asterisk instances on both Debian and > CentOS based VPS's, and got them working fine. However, I recently > installed a copy of Astlinux and installed on a WRAP board and I'm > totally stuck! > > I'm using sipgate.co.uk for incoming calls, but when I make a test > call from the PSTN, the call just dies without connecting to my > Astlinux box. (I'm monitoring asterisk console via 'asterisk -rv' > and see nothing). > > I wondered if it might be a problem with Asterisk not listening > properly, or perhaps a problem with my home firewall. Would anyone be > kind enough to advise me as to where I may have gone wrong? > > Thanks, Chris. > > My sip.conf looks like this: > > register => 277:[EMAIL PROTECTED]/277 You should use ngrep when making a call to see what is happening on the wire. You don't mention whether or not you can make outbound calls so I will ask now, can you make outbound calls? What do you see on a sip show peer 277 is your line registered. I'm unsure about Astlinux but if you've seen em one you've seen em all, is iptables running on the machine itself (iptables -L), is the device connected properly, can it reach other places say Google. Are you doing NAT if so, did you configure a STUN server, did you specific NAT in sip.conf. In a terminal on the configured box - run the following: ngrep -d YOUR_ETHERNET_CARD 227 udp port 5060 Place a call, what do you see, if nothing comes through (these are SIP messages by the way) then its not hitting your machine period, whether its a firewall, ACL on a router, doesn't make a difference, its not hitting the box, you have to troubleshoot from there. Starting point, throw the box in a DMZ, with the same ngrep command, place another call, technically you should see some messages hitting the machine. Have you contacted your provider, are they doing any kind of IP address filtering, SIP filtering. -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo SGFA #579 (FW+VPN v4.1) SGFE #574 (FW+VPN v4.1) CEH/CNDA, CHFI "Experience hath shewn, that even under the best forms (of government) those entrusted with power have, in time, and by slow operations, perverted it into tyranny." Thomas Jefferson wget -qO - www.infiltrated.net/sig|perl http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x3AC173DB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming call does not reach asterisk.
Hello, >From the netstat output my initial *guess* is that asterisk is listening (udp/5060, udp/2727, among others). One way to tell for sure would be to run 'lsof -i' which would show you the process associated with the port. As far as the call not reaching asterisk or being a firewall issue, one way to tell might be to start a tcpdump just prior to making the incoming call. Something like this: tcpdump -n 'port 5060' That would show the connection attempt. But if it's not showing up on the console then chances are that it's not even reaching the asterisk server to begin with. Steve On Sun, Jul 13, 2008 at 12:03:29AM +0100, Chris Rowson wrote: > Hi, this is my first post to the list, but I have tried to search > elsewhere for a solution, and have had a read of 'Asterisk - The > Future of Telephony'. So you could say that I have at least tried to > RTFM as it were! > > I've configured a couple of Asterisk instances on both Debian and > CentOS based VPS's, and got them working fine. However, I recently > installed a copy of Astlinux and installed on a WRAP board and I'm > totally stuck! > > I'm using sipgate.co.uk for incoming calls, but when I make a test > call from the PSTN, the call just dies without connecting to my > Astlinux box. (I'm monitoring asterisk console via 'asterisk -rv' > and see nothing). > > I wondered if it might be a problem with Asterisk not listening > properly, or perhaps a problem with my home firewall. Would anyone be > kind enough to advise me as to where I may have gone wrong? > > Thanks, Chris. > > My sip.conf looks like this: > > -- > [general] > context = default ;default context for incoming calls > bindport = 5060 > bindaddr = 0.0.0.0 > srvlookup = yes > disallow=all;disallow all codecs > allow=alaw ;except alaw (1st pref) > allow=ulaw ;and ulaw (second pref) > > register => 277:[EMAIL PROTECTED]/277 > > [sipgate] ;sipgate sip in on 01482 77 > type=peer > context=from-pots > fromuser=277 > username=277 > authuser=277 > secret=*** > host=sipgate.co.uk > fromdomain=sipgate.co.uk > dtmfmode=inband > insecure=very > canreinvite=no > disallow=all > allow=alaw > allow=ulaw > nat=yes > qualify=yes > - > My extensions.conf looks like this: > > - > [general] > static=yes > writeprotect=np > autofallthrough=yes > clearglobalvars=no > priorityjumping=no > > [from-pots] > exten => s,1,Answer() > exten => s,n,Wait(3) > exten => s,n,Playback(tt-weasels) > exten => s,n,Hangup() > -- > > and netstat looks like this > > -- > Active Internet connections (only servers) > Proto Recv-Q Send-Q Local Address Foreign Address State > tcp0 0 *:www *:* LISTEN > tcp0 0 *:ftp *:* LISTEN > tcp0 0 *:ssh *:* LISTEN > tcp0 0 *:https *:* LISTEN > udp0 0 *:1025 *:* > udp0 0 *:1026 *:* > udp0 0 *:1027 *:* > udp0 0 *:1028 *:* > udp0 0 *:1029 *:* > udp0 0 *:1030 *:* > udp0 0 *:1031 *:* > udp0 0 *:1032 *:* > udp0 0 *:2727 *:* > udp0 0 *:4520 *:* > udp0 0 *:5060 *:* > udp0 0 *:tftp *:* > udp0 0 *:4569 *:* > udp0 0 *:5353 *:* > udp0 0 *:5353 *:* > udp0 0 *:5353 *:* > udp0 0 *:5353 *:* > udp0 0 *:5353 *:* > udp0 0 *:5353 *:* > udp0 0 *:5353 *:* > udp0 0 *:5353 *:* > udp0 0 *:ntp *:* > - > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-us
Re: [asterisk-users] Tracking Call Time While in Dial()
On Sat, 12 Jul 2008, Douglas Garstang wrote:
> The person I am working is building a calling card. They want to allow
> the user to recharge their account when their time runs out (without
> hanging up the current call). I got no idea how to implement that. In
> addition, they don't want to charge the user for the time they spend
> recharging their account. So, they need to track multiple timers for the
> call.
Would this work?
Hack up meetme to add an option to time out after x seconds, set
MEETME_STATUS to "TIMEOUT" and return to the dialplan. You can follow the
"w" option to figure out how to pass the seconds. Then in meetme's main
loop (in conf_run()), just check for when its time to leave.
Use dial()'s G option to put both the caller (with the timeout) and the
callee into a meetme.
When the caller times out, the callee will hear the music on hold ("please
wait until the other party returns").
The caller can top up and then return to the meetme. You can accumulate
the time the caller is in the meetme.
What happens if the caller can't top up and hangs up. Who pays for the
callee leg? Would you need a second timer to abort the callee?
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Incoming call does not reach asterisk.
Hi, this is my first post to the list, but I have tried to search elsewhere for a solution, and have had a read of 'Asterisk - The Future of Telephony'. So you could say that I have at least tried to RTFM as it were! I've configured a couple of Asterisk instances on both Debian and CentOS based VPS's, and got them working fine. However, I recently installed a copy of Astlinux and installed on a WRAP board and I'm totally stuck! I'm using sipgate.co.uk for incoming calls, but when I make a test call from the PSTN, the call just dies without connecting to my Astlinux box. (I'm monitoring asterisk console via 'asterisk -rv' and see nothing). I wondered if it might be a problem with Asterisk not listening properly, or perhaps a problem with my home firewall. Would anyone be kind enough to advise me as to where I may have gone wrong? Thanks, Chris. My sip.conf looks like this: -- [general] context = default ;default context for incoming calls bindport = 5060 bindaddr = 0.0.0.0 srvlookup = yes disallow=all;disallow all codecs allow=alaw ;except alaw (1st pref) allow=ulaw ;and ulaw (second pref) register => 277:[EMAIL PROTECTED]/277 [sipgate] ;sipgate sip in on 01482 77 type=peer context=from-pots fromuser=277 username=277 authuser=277 secret=*** host=sipgate.co.uk fromdomain=sipgate.co.uk dtmfmode=inband insecure=very canreinvite=no disallow=all allow=alaw allow=ulaw nat=yes qualify=yes - My extensions.conf looks like this: - [general] static=yes writeprotect=np autofallthrough=yes clearglobalvars=no priorityjumping=no [from-pots] exten => s,1,Answer() exten => s,n,Wait(3) exten => s,n,Playback(tt-weasels) exten => s,n,Hangup() -- and netstat looks like this -- Active Internet connections (only servers) Proto Recv-Q Send-Q Local Address Foreign Address State tcp0 0 *:www *:* LISTEN tcp0 0 *:ftp *:* LISTEN tcp0 0 *:ssh *:* LISTEN tcp0 0 *:https *:* LISTEN udp0 0 *:1025 *:* udp0 0 *:1026 *:* udp0 0 *:1027 *:* udp0 0 *:1028 *:* udp0 0 *:1029 *:* udp0 0 *:1030 *:* udp0 0 *:1031 *:* udp0 0 *:1032 *:* udp0 0 *:2727 *:* udp0 0 *:4520 *:* udp0 0 *:5060 *:* udp0 0 *:tftp *:* udp0 0 *:4569 *:* udp0 0 *:5353 *:* udp0 0 *:5353 *:* udp0 0 *:5353 *:* udp0 0 *:5353 *:* udp0 0 *:5353 *:* udp0 0 *:5353 *:* udp0 0 *:5353 *:* udp0 0 *:5353 *:* udp0 0 *:ntp *:* - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bridging two Redirected Channels?
All, I was able to use the Redirect AMI command to take two bridged channels and send them elsewhere in the dial plan. Great. Now... how can I bridge them back together again? Looks like Asterisk 1.6 might have a bridge command. What about Asterisk 1.4? Doug. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wanted Polycom 601 + expansion sidecar
If anyone on the list has a Polycom 601 + sidecar expansion they want to sell for less than $250 including shipping to New York 10027 then email me details. Cheers, Dean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tracking Call Time While in Dial()
The person I am working is building a calling card. They want to allow the user to recharge their account when their time runs out (without hanging up the current call). I got no idea how to implement that. In addition, they don't want to charge the user for the time they spend recharging their account. So, they need to track multiple timers for the call. Doug. - Original Message From: Tzafrir Cohen <[EMAIL PROTECTED]> To: asterisk-users@lists.digium.com Sent: Saturday, July 12, 2008 1:46:13 AM Subject: Re: [asterisk-users] Tracking Call Time While in Dial() On Fri, Jul 11, 2008 at 10:52:53AM -0700, Douglas Garstang wrote: > Wanting to provide a real time call timer on a web page. Can't you get information about other channels through the manager interface without this special AGI? Maybe you just need to somehow mark those channels as "interesting" before the Dial, or write out start time to a variable before the Dial starts. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor fdiles
Hello, I am running Asterisk 1.4.20-1 and having the exact same problem. It looks like others are having issues as well according to this thread: http://www.trixbox.org/forums/trixbox-forums/help/recordings-out-sync-using-mixmonitor Anyone have any idea's? On Wed, Apr 9, 2008 at 7:16 AM, robert boardman <[EMAIL PROTECTED]> wrote: > Hi, > > I have a load of files recorded with MixMonitor that are out of sync ie > one leg of the call is 2-3 seconds behind the other, > > is this a bug in Asterisk 1.4.18, or am I possibly doing something wrong > > > Is it possible to edit the file and re sync the a/b leg? > > Thanks for your help > > Robb > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Isaac McDonald Got VoIP? [EMAIL PROTECTED] Cell: +1 253-223-8673 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
On Sat, Jul 12, 2008 at 11:52:21AM -0400, Steve Totaro wrote: > On Sat, Jul 12, 2008 at 11:18 AM, Tzafrir Cohen > <[EMAIL PROTECTED]> wrote: > > On Sat, Jul 12, 2008 at 09:54:37AM -0400, Steve Totaro wrote: > >> On Sat, Jul 12, 2008 at 9:32 AM, Michael Graves <[EMAIL PROTECTED]> wrote: > >> > On Fri, 11 Jul 2008 19:26:06 -0400, Steve Totaro wrote: > >> > > >> >>As Michael Graves points out, people will hack it to run on thin > >> >>clients and why not virtual machines with very limited access? Maybe > >> >>an AP with a USB port and OpenWRT or something? > >> > > >> > Since it needs to run their app it's probablly limited to x86 right > >> > now. Thin clients running XPe are suitable hosts, probably not routers. > >> > Not open source so a port is unlikely. > >> > >> I read on one page that linux support was to come later "this year". > >> > >> Do you know how many times I have seen or heard "An Open Source Port > >> is not Likely"? > >> > >> Who says it "must" run their app? Can it run in Wine? Where there is > >> a will there is a way. Sniff enough packets and with some mojo, > >> anything can be emulated on the wire. > > > > What is their network different than some random SIP provider? Besides > > the fact that you can't just call a random SIP provider? > > > > Add that to the fact that it won't "just work". > > > > I don't have one but I assume it is just a mass produced commodity > sound card with some flash to store their software and configs. There are cheaper USB sound cards. Any USB "voip phone" / "skype phone" is (also) a USB sound device. This is a standard class of USB devices, and is well supported (at least in Linux). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
On Sat, Jul 12, 2008 at 11:18 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > On Sat, Jul 12, 2008 at 09:54:37AM -0400, Steve Totaro wrote: >> On Sat, Jul 12, 2008 at 9:32 AM, Michael Graves <[EMAIL PROTECTED]> wrote: >> > On Fri, 11 Jul 2008 19:26:06 -0400, Steve Totaro wrote: >> > >> >>As Michael Graves points out, people will hack it to run on thin >> >>clients and why not virtual machines with very limited access? Maybe >> >>an AP with a USB port and OpenWRT or something? >> > >> > Since it needs to run their app it's probablly limited to x86 right >> > now. Thin clients running XPe are suitable hosts, probably not routers. >> > Not open source so a port is unlikely. >> >> I read on one page that linux support was to come later "this year". >> >> Do you know how many times I have seen or heard "An Open Source Port >> is not Likely"? >> >> Who says it "must" run their app? Can it run in Wine? Where there is >> a will there is a way. Sniff enough packets and with some mojo, >> anything can be emulated on the wire. > > What is their network different than some random SIP provider? Besides > the fact that you can't just call a random SIP provider? > > Add that to the fact that it won't "just work". > I don't have one but I assume it is just a mass produced commodity sound card with some flash to store their software and configs. Thanks, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
On Sat, Jul 12, 2008 at 09:54:37AM -0400, Steve Totaro wrote: > On Sat, Jul 12, 2008 at 9:32 AM, Michael Graves <[EMAIL PROTECTED]> wrote: > > On Fri, 11 Jul 2008 19:26:06 -0400, Steve Totaro wrote: > > > >>As Michael Graves points out, people will hack it to run on thin > >>clients and why not virtual machines with very limited access? Maybe > >>an AP with a USB port and OpenWRT or something? > > > > Since it needs to run their app it's probablly limited to x86 right > > now. Thin clients running XPe are suitable hosts, probably not routers. > > Not open source so a port is unlikely. > > I read on one page that linux support was to come later "this year". > > Do you know how many times I have seen or heard "An Open Source Port > is not Likely"? > > Who says it "must" run their app? Can it run in Wine? Where there is > a will there is a way. Sniff enough packets and with some mojo, > anything can be emulated on the wire. What is their network different than some random SIP provider? Besides the fact that you can't just call a random SIP provider? Add that to the fact that it won't "just work". -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack and Skype call quality
Jason Aarons (US) wrote: > My understanding is Skype's secret is using the iLBC codec, which Cisco > has also licensed for their 79X2 models as well. I travel and lot and > in places where Yahoo Phone Out or MSN Phone or Cisco IP Communicator > will fail the Skype client will work. The iLBC codec can really handle > packet loss. AFAIK, not iLBC, but another GIPS codec. ILBC is present in some grandstream phones from what I remember, not in the Cisco 7912 as one example, not sure about other phones. In a word with still many PSTN gateways out there that don't support it, ILBC in the SIP UA side only can help that much :-).. > > Skype High Quality Video with the Logitech Orbit AF on both ends is > awesome. I got my family a set for Fathers day. Just amazing video > quality. Uses a On2 VP-7 codec that has much lower cpu and other > benefits over h.264. > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Steve > Underwood > Sent: Saturday, July 12, 2008 3:30 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] MagicJack quality > > Tzafrir Cohen wrote: >> On Sat, Jul 12, 2008 at 10:26:24AM +0800, Steve Underwood wrote: >> >>> C. Savinovich wrote: >>> I am puzzled by the quality of magicjack. I keep trying to figure > out how they can the quality be that adequate. Since Skype also has an > excellent quality, that leaves me to believe that software based calls > (softphones) could have and advantage over hardphones, provided there is a > parameter that those 2 companies are addressing. Anyone's thoughts on this? CS >>> I don't know what Magic-jack does (I've never actually seen one), but > I >>> know the key thing about Skype that impresses people - its wideband >>> voice codec. A lot of people poo-poo the idea that wideband voice has > >>> value in a phone call. They are either close to deaf, or have never >>> tried it. Clarity is profoundly improved. Skype seems to use various >>> tricks to keep the packet flow smooth, but its wideband that makes it > >>> sound better than the PSTN. >>> >>> You might think a standard phone plugged into an adaptor, like a >>> Magic-jack, would be limited to narrow band voice, as that is all the > >>> phone was designed for. It turns out most phones only aggressively >>> filter at the low end of the band. They let a lot of energy above > 4kHz >>> through, and they do generally sound better through a wideband codec. >>> >>> Many modern line interface chips are actually capable of running in a > >>> 16k samples/second mode, even though most are programmed for 8k >>> samples/second. I think the ones on the TDM400P type cards can. Some >>> from Silicon Labs certainly can, and chips from Zarlink and others > can. >>> >> The DAA in those cards can work in 16Hz. So they can send higher > quality >> samples to the telco. Provided Zaptel supports it. But then again, it >> will get lost as soon as it gets converted to digital at the telco, >> right? >> > I guess I wasn't clear. What I said was only useful for a SLIC to phone > connection. It won't be of any benefit for a DAA to PSTN exchange > connection, for the reason you state. >> Anyway, the ProSLIC chip does not seem to support it. >> > Silicon Labs make a Wideband ProSLIC, Si 3216, which is, er, wideband. > As I said before, Zarlink and other make them too. > > Regards, > Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack and Skype call quality
My understanding is Skype's secret is using the iLBC codec, which Cisco has also licensed for their 79X2 models as well. I travel and lot and in places where Yahoo Phone Out or MSN Phone or Cisco IP Communicator will fail the Skype client will work. The iLBC codec can really handle packet loss. Skype High Quality Video with the Logitech Orbit AF on both ends is awesome. I got my family a set for Fathers day. Just amazing video quality. Uses a On2 VP-7 codec that has much lower cpu and other benefits over h.264. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Saturday, July 12, 2008 3:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MagicJack quality Tzafrir Cohen wrote: > On Sat, Jul 12, 2008 at 10:26:24AM +0800, Steve Underwood wrote: > >> C. Savinovich wrote: >> >>> I am puzzled by the quality of magicjack. I keep trying to figure out how >>> they can the quality be that adequate. Since Skype also has an excellent >>> quality, that leaves me to believe that software based calls (softphones) >>> could have and advantage over hardphones, provided there is a parameter that >>> those 2 companies are addressing. >>> >>> Anyone's thoughts on this? >>> >>> CS >>> >>> >> I don't know what Magic-jack does (I've never actually seen one), but I >> know the key thing about Skype that impresses people - its wideband >> voice codec. A lot of people poo-poo the idea that wideband voice has >> value in a phone call. They are either close to deaf, or have never >> tried it. Clarity is profoundly improved. Skype seems to use various >> tricks to keep the packet flow smooth, but its wideband that makes it >> sound better than the PSTN. >> >> You might think a standard phone plugged into an adaptor, like a >> Magic-jack, would be limited to narrow band voice, as that is all the >> phone was designed for. It turns out most phones only aggressively >> filter at the low end of the band. They let a lot of energy above 4kHz >> through, and they do generally sound better through a wideband codec. >> >> Many modern line interface chips are actually capable of running in a >> 16k samples/second mode, even though most are programmed for 8k >> samples/second. I think the ones on the TDM400P type cards can. Some >> from Silicon Labs certainly can, and chips from Zarlink and others can. >> > > The DAA in those cards can work in 16Hz. So they can send higher quality > samples to the telco. Provided Zaptel supports it. But then again, it > will get lost as soon as it gets converted to digital at the telco, > right? > I guess I wasn't clear. What I said was only useful for a SLIC to phone connection. It won't be of any benefit for a DAA to PSTN exchange connection, for the reason you state. > Anyway, the ProSLIC chip does not seem to support it. > Silicon Labs make a Wideband ProSLIC, Si 3216, which is, er, wideband. As I said before, Zarlink and other make them too. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AsteriskNow SIP config
I can not seem to get AsteriskNow to register my SIP provider correctly? I can do this manually when compiling Asterisk and installing it w/o a GUI, but not with this. I just get the following message. -- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #22) The register line I use normally looks like: user:[EMAIL PROTECTED]:port but the above looks simplified? Is that only a result of what the logging looks like? Any ideas? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
On Sat, Jul 12, 2008 at 9:32 AM, Michael Graves <[EMAIL PROTECTED]> wrote: > On Fri, 11 Jul 2008 19:26:06 -0400, Steve Totaro wrote: > >>As Michael Graves points out, people will hack it to run on thin >>clients and why not virtual machines with very limited access? Maybe >>an AP with a USB port and OpenWRT or something? > > Since it needs to run their app it's probablly limited to x86 right > now. Thin clients running XPe are suitable hosts, probably not routers. > Not open source so a port is unlikely. > > Michael > > -- > Michael Graves > mgravesmstvp.com > http://blog.mgraves.org > o713-861-4005 > c713-201-1262 > sip:[EMAIL PROTECTED] > skype mjgraves > [EMAIL PROTECTED] > I read on one page that linux support was to come later "this year". Do you know how many times I have seen or heard "An Open Source Port is not Likely"? Who says it "must" run their app? Can it run in Wine? Where there is a will there is a way. Sniff enough packets and with some mojo, anything can be emulated on the wire. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
On Fri, 11 Jul 2008 19:26:06 -0400, Steve Totaro wrote: >As Michael Graves points out, people will hack it to run on thin >clients and why not virtual machines with very limited access? Maybe >an AP with a USB port and OpenWRT or something? Since it needs to run their app it's probablly limited to x86 right now. Thin clients running XPe are suitable hosts, probably not routers. Not open source so a port is unlikely. Michael -- Michael Graves mgravesmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura 3000 replacement ---> SPA3102 how reliable is it?
Hans Witvliet wrote: > There's not much that can stand lightning (not just a direct hit), so > you cant't blame the sipura box for that. > Even when it was build, using a Faraday-cage with double insulation with > optocouplers, the amount of energy picked up by a 3 km line is beyond > commercial engineerd products. So I've found out. We moved here exactly a year ago. Up until April everything was OK, That's after I forced the telco to virtually recable the whole run, (a neighbour likes to take pot shots at road signs etc. they actually found pellets in the cable) April we had a sequence of very violent thunderstorms, the first took out a Sipura 3102, a Linksys 8 port switch and the POE etc of an Aastra 9133i. The shock was so strong it caused the cutout at the meter to drop out, that's 200mtrs away. Exactly one week later a repeat performance, but this time I'd not replaced the Aastra. The third time I woke up to an enormous crash of thunder to see that the power had gone again, I was at the door of the office when the second strike occurred and actually saw the flash as the remaining SPA3000 I had grilled. When the electrician inspected the circuits the next day he told me the previous occupants had lost telephones nearly every time there was a big storm. What is interesting is that crappy little handsets which take their power from the phone line survive and an ADSL modem on the same line has also survived. Who said lightening never strikes twice. DC ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tracking Call Time While in Dial()
On Fri, Jul 11, 2008 at 10:52:53AM -0700, Douglas Garstang wrote: > Wanting to provide a real time call timer on a web page. Can't you get information about other channels through the manager interface without this special AGI? Maybe you just need to somehow mark those channels as "interesting" before the Dial, or write out start time to a variable before the Dial starts. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sipura 3000 replacement ---> SPA3102 how reliable is it?
On Fri, 2008-07-11 at 18:37 +0200, Dave Cotton wrote: > SIP wrote: > > Joseph wrote: > >> I need another Sipura 3K and the replacement I think is Linksys SPA3102. > >> Any input on how reliable is it? > >> > >> > > We have a few dozen subscribers using them at any given point in time. I > > and my wife even use them at our respective homes. Rock solid stable. > > No issues whatsoever. > > The only reservation I've got with the 3000/3102 units is that I've had > 3 destroyed by lightening recently. But I'm told it's because I'm on the > end of 3kms of cable across open countryside. The others I've installed > in non rural installations work faultlessly. > There's not much that can stand lightning (not just a direct hit), so you cant't blame the sipura box for that. Even when it was build, using a Faraday-cage with double insulation with optocouplers, the amount of energy picked up by a 3 km line is beyond commercial engineerd products. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming
On Fri, 2008-07-11 at 11:22 -0500, Tilghman Lesher wrote: > On Friday 11 July 2008 09:17:37 Artie Gold wrote: > > In updating to 1.4.21 recently, we've encountered a problem, when running > > over a satellite connection (where the latency is considerable; a "regular" > > internet connection did not exhibit this problem), where incoming calls are > > being dropped as a result of the sip handshake timing out (dropping down to > > 1.4.18.1 solved the problem for us). From reading the change logs and other > > posts, it seems that some work has been done in this area recently to get > > it "right"; it appears that, at least in the satellite case, things may > > have gotten a little too "tight"... > > > > If this rings a bell for anyone, any insight would be appreciated. > > Try setting t1min to something higher than the default, 100 (ms). This value > is settable globally, as well as per-peer. > I've encoutred latencies about 600ms, so timeout of 100 ms is abit short. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MagicJack quality
Tzafrir Cohen wrote: > On Sat, Jul 12, 2008 at 10:26:24AM +0800, Steve Underwood wrote: > >> C. Savinovich wrote: >> >>> I am puzzled by the quality of magicjack. I keep trying to figure out how >>> they can the quality be that adequate. Since Skype also has an excellent >>> quality, that leaves me to believe that software based calls (softphones) >>> could have and advantage over hardphones, provided there is a parameter that >>> those 2 companies are addressing. >>> >>> Anyone's thoughts on this? >>> >>> CS >>> >>> >> I don't know what Magic-jack does (I've never actually seen one), but I >> know the key thing about Skype that impresses people - its wideband >> voice codec. A lot of people poo-poo the idea that wideband voice has >> value in a phone call. They are either close to deaf, or have never >> tried it. Clarity is profoundly improved. Skype seems to use various >> tricks to keep the packet flow smooth, but its wideband that makes it >> sound better than the PSTN. >> >> You might think a standard phone plugged into an adaptor, like a >> Magic-jack, would be limited to narrow band voice, as that is all the >> phone was designed for. It turns out most phones only aggressively >> filter at the low end of the band. They let a lot of energy above 4kHz >> through, and they do generally sound better through a wideband codec. >> >> Many modern line interface chips are actually capable of running in a >> 16k samples/second mode, even though most are programmed for 8k >> samples/second. I think the ones on the TDM400P type cards can. Some >> from Silicon Labs certainly can, and chips from Zarlink and others can. >> > > The DAA in those cards can work in 16Hz. So they can send higher quality > samples to the telco. Provided Zaptel supports it. But then again, it > will get lost as soon as it gets converted to digital at the telco, > right? > I guess I wasn't clear. What I said was only useful for a SLIC to phone connection. It won't be of any benefit for a DAA to PSTN exchange connection, for the reason you state. > Anyway, the ProSLIC chip does not seem to support it. > Silicon Labs make a Wideband ProSLIC, Si 3216, which is, er, wideband. As I said before, Zarlink and other make them too. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
