Re: [Asterisk-Users] Asterisk on sattelite link
marius baranescu <[EMAIL PROTECTED]> writes: > I have a running Asterisk box . It is running great > My problem is that I can not get connected to the world :) . > My only option available here is a satellite connection . > I was testing different service providers but all of them are doing > firewalling and NAT so SIP, IAX are not working > I desperately need to get connected to the world :)) > Please recommend me a good ISP for Middle East (permanent 2 way > connection) , real IP adresses etc I guess it depends on where you are in the Middle East. My experience is in Iraq. In this region are lots of really poor providers reselling highly unreliable, congested, and jitter-prone bandwidth. There are a few that are not. In major cities in Iraq TigrisNet resells Intelsat bandwidth via a metro 802.11a network. Elsewhere in Iraq (and likely the whole Middle East) you can buy service from Intelsat directly. Its not cheap, but it works very well. -- Matt Ranney - [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + Satellite connection
"Jay Milk" <[EMAIL PROTECTED]> writes: > There have been prior discussion on this on the list -- just google for > "starband site:lists.digium.com" and you should find it. IIRC, there > has been some sporadic success, but overwhelmingly, satellite-based VOIP > connections are not considered feasible. My company does IAX2/GSM over satellite links for many sites throughout the world on through different satellite providers. It is fairly reliable. There is obviously a lot of noticeable lag with 700ms RTTs, but you get used to that right away. As long as there is a guarantee as to the minimum amount of bandwidth your provider will give you, VoIP is feasible over satellite links. Once the next generation of jitter buffer, timing, etc. gets finished in asterisk, things will get even better over sat links and less than desirable WAN paths. -- Matt Ranney - [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco poe + netgear
Jeb Campbell <[EMAIL PROTECTED]> writes: > By far the best poe (price/performance) I have seen for Cisco poe (or > standard poe) is the Netgear > FSM7326P. http://www.cdw.com/shop/products/default.aspx?EDC=568864 > > It is a managed layer3 poe switch (24 port) with 2 gigabit ports also. > > Works out of the box with Cisco and Snoms (it auto detects which > polarity they want). No adapters needed for either. And it is about > $1100. > > We are using 4 of them and love them. We are using 9 of them and are pretty unhappy with them. Its really nice that they power the Cisco phones, and they are very cheap. However, we've had about one "lockup" per month with these switches where the only solution to pass traffic again was to power cycle the unit. -- Matt Ranney - [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 76XX - How to ignore a call (silence ring)
"Shawn Kelley" <[EMAIL PROTECTED]> writes: > I am preparing to setup a system using Cisco 7940 and 7960's I have the > 7.1 SIP firmware on them. > One issue I have run into is how to silence the ringer if a call comes > in and you don't want to take it. > Many phones have a DND button. I know the 79XX has the DND in the menu > but it is to cumbersome to go into the settings then phone preferences > then the DND and select yes. > Is there any other way this can be achieved to silence the ringer if a > call comes in and the person doesn't want to take it? I've asked the Cisco TAC about this, and they told me I needed to submit a "feature request" with my VAR. If you bought your phones through a Cisco reseller with some clout, I'd encourage you to submit a feature request for DND that does what DND does on all other phone systems. -- Matt Ranney - [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 79xx + asterisk + some functions Q
Alex Ongena <[EMAIL PROTECTED]> writes: > 1) Status info: can I see on my 7960 equipment (eventualy >with the 7914 extension) who is free/busy and alike ? While it looks like recent SIP images support this "multiple call appearance" feature, to my knowledge asterisk does not. -- Matt Ranney - [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call quality monitoring
Chris Icide <[EMAIL PROTECTED]> writes: > Satellite links can be pretty tough to troubleshoot. It sounds like > you are running into a uplink buffer issue. On heavily loaded > uplinks, the input buffers can get quite large, and if the satellite > provider isn't using some form of buffer handling that prioritizes udp > traffic, it may be that most of your voice packets are falling on the > floor of the uplink facility... Yeah, I thought about that too, which is why I set up tools to monitor packet loss and measure jitter. It uses a non-conflicting UDP port near the one that IAX2 uses. My tests indicate very little loss and the same jitter in both directions. What I really want is a way to get asterisk to yelp if it notices that its about to make the call sound bad due to late/missing packets or for whatever reason. Right now it seems that the network is functioning normally but still one direction of the calls sounds intermittently awful. -- Matt Ranney - [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call quality monitoring
I need to debug a call quality issue with remote users on the other end of a satellite link. The symptoms are: we here on the Internet side can hear them just fine. On their end, things work sorta OK most times, but they often suffer from severe dropouts and digital warbling, both of which I attribute to them missing packets. Often times they can't make out a word we are saying while we can hear them crystal clearly. Various pings and other network tests indicate that the underlying network is functioning as well as can be expected for a sat link. In fact, the overall jitter seems to be pretty low (avg 20ms). Packet loss is around 1-2%, and latency is around 700ms on average. I'm left to assume that the jitter buffer on that end isn't functioning properly. Both ends of the call have the same jitter buffer settings. The call is carried by IAX2 and encoded with ILBC. The iax.conf files on each end start like this: >[general] >trunk=no >notransfer=yes >iaxcompat=no > >bandwidth=low > >disallow=all >allow=ilbc > >jitterbuffer=yes >dropcount=3 >maxjitterbuffer=500 >maxexcessbuffer=150 >minexcessbuffer=40 >jittershrinkrate=1 Of course, perhaps the jitter buffer isn't to blame, but given that one side of the call sounds perfect, I can't think of anything else obvious that would cause this. Is there any way to extract from asterisk some idea of why it thinks the calls sound bad? For example, when the jitter buffer notices that packets are discarded because they are too late, when excessive packets are completely missing, etc. I've been collecting a giant debug log for a while now, so I could pretty easily sift through it if there's something good to look for. Thanks. -- Matt Ranney - [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
