Re: [asterisk-users] VoIP over virtualized VPN
I use openvpn for VOIP traffic all the time. It's not a commercial application, and only one simultaneous call usually on each vpn link, but I even have a VPN client on a Linksys WRT-54g wireless router with 1 phone behind it - it works flawlessly, so it does not take a lot of CPU to run a vpn connection. Andrew 2010/5/26 Motiejus Jakštys : > Hi List, > Our company has several small distributed offices we would like to > inter-connect with bridged VPN a single subnet (last example in > http://www.shorewall.net/OPENVPN.html). We have SIP phones in every > office (up to 5) so we can use SIP without any NATing and securely. > Max theoretical simultaneous calls possible ~30, but we have ~5-10 @ > regular basis. > OpenVPN server would be in the same datacenter like Asterisk PBX (in > one physical subnet). Asterisk and OpenVPN are virtualized XEN guests. > > I wonder about overheads, system loads and other possible gotchas in > this setup. Is there anything I should (re-)consider before > implementing this? Anyone had difficulties running VoIP or VPN traffic > over (virtualized if it makes any difference) VPN? > We use mainly g729 and speex, and very little g711. > > Regards > Motiejus > > -- > _ > -- 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 > -- _ -- 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
Re: [asterisk-users] G.711a or G.711u ???
On Tue, Mar 23, 2010 at 1:41 PM, Alejandro Cabrera Obed wrote: > Dear all, I have an Asterisk SIP server in a LAN environment and I want your > opinion in order to decide the use of an audio codec: > > What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip > calls ??? As others have stated, use whatever matches your country. In terms of the network, the traffic is exactly the same for both a-law and u-law. Both are 8khz sampled 8 bits per sample PCM (64kbps). The only difference between the two is the companding (http://en.wikipedia.org/wiki/Companding) algorithm that's used for the audio emphasis. To the network, the packets are 100% identical. Andrew > Thank you !!! > > Alejandro > -- _ -- 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
Re: [asterisk-users] SIP tunnel
On Thu, Feb 11, 2010 at 6:37 AM, mosbah.abdelkader wrote: > Hello, > > I have the following situation: A firewall is blocking all SIP and RTP > traffic in the side of some of my clients. My clients cannot change settings > of the firewall. > > I need to solve this problem and I need some help from you. > I would definitely say use a VPN. All you need is one UDP port accessible on the server side (and no outgoing connection blocks on the firewalled side, which is usually the case - at least something has to be open somewhere), and then you can run any protocol you want, that uses any ports, and no problem at all. Check out OpenVPN. It's free, easy to setup, and has clients for all platforms. Andrew -- _ -- 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
Re: [asterisk-users] odd issue with the with SIP over VPN
You probably are not advertising the routes across the vpn properly. Does your setup look like this asterisk[network a]openVPN server[network b - vpn]-openVPN client[network c]-sip client where network a, b, and c are all separate subnets? Is your vpn setup for routing or bridging? you need to make sure that the vpn server allows network a to to talk to network c, and that network c can talk to network a. By default, only the client can talk to the server. Attached subnets will not be routed automatically. See this section of the openVPN howto: http://openvpn.net/index.php/open-source/documentation/howto.html#scope I have 3 asterisk servers with sip trunking between them all running over openVPN links, and everything works fine when you make sure you setup the routing right in the vpn. I also have 2 phones that connect to one of the servers over a openVPN link as well - they're not sip (Nortel unistim) but it also works just fine. Andrew On Sat, Jan 23, 2010 at 7:17 PM, Alex Balashov wrote: > You're going to be a lot more specific about the precise - if > symptomatic - meaning of "do not." > > On 01/23/2010 09:08 PM, Zane C.B. wrote: > >> I've run into a odd issue where inbound calls to the SIP client work >> fine, but outbound from the SIP client do not. >> >> The path between the client and the server is as below. >> >> N900 SIP client<-- OpenVPN --> Asterisk >> >> The version of Asterisk in question is 1.6.0.18. >> >> Any suggestions? >> > > > -- > Alex Balashov - Principal > Evariste Systems LLC > > Tel : +1 678-954-0670 > Direct : +1 678-954-0671 > Web : http://www.evaristesys.com/ > > -- > _ > -- 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 > -- _ -- 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
Re: [asterisk-users] 10/100 voip phones and gigabit connection
Windows, yes, but used to be through 3rd party software. Doubt this has changed as Windows has no focus on any useful network anything. Linux, yes, and it's definitely not complicated. Probably take 2 minutes to setup if you already had bridge utils installed, maybe 5 if you had to install the package first. Regardless, I would say this wouldn't be the best solution for the reasons someone already mentioned - you form some dependency between the phone and the PC. 2 cables is definitely the best, followed by a cheap gig switch at each desk. Also, someone was mentioning if you could run 2 ethernet connections through one cable. This works with 10/100 (as only 2 pairs are used, so you can wire 2 pairs to one jack, and the other 2 pairs in the cat5 to the other jack), but doesn't work with gigabit, as it uses all 4 pairs. Even if you just consider 10/100, this is a nifty hack in a pinch, but I seriously doubt anyone does this in a professional install. Cable isn't all that expensive when it comes right down to it. Andrew On Fri, Jan 15, 2010 at 6:38 PM, Jeff LaCoursiere wrote: > > > On Fri, 15 Jan 2010, Hans Witvliet wrote: > >> >> If you connect your pc with GB-lan card to an dual-ported ip-phone, you >> and up with an 100Mbps lan connection to your pc. >> >> Only way to avoid that, is to insert a cheap second lan-card in your pc, >> and connect your phone to the second lan, so your pc will act as an >> switch, instead of your phone... > > I'm curious - how have you managed to connect a second LAN card and have > it bridge your (presumably onboard) ethernet? Does Windows have such > capability? But I guess the OP was running XUbuntu, and though relatively > complicated I guess you could get it to do that. > > j > > -- > _ > -- 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 > -- _ -- 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
Re: [asterisk-users] SIP over VPN -- no audio to other remote/VPNconnected phones
Are you using openvpn? If so, there's an option in the server config file that allows vpn clients to talk to other vpn clients, otherwise they can only talk to the server. Using canreinvite=no is just forcing the traffic to go through the server, which is why that makes it work. I must say VPN + VOIP is an extremely powerful combination! Andrew On Mon, Jan 11, 2010 at 12:39 PM, Ryan McCormack wrote: > canreinvite=no did the trick! Thanks!! >> [Cary Fitch] >> >> One thought: if you are using "reinvite" try turning that off. That will be >> a clue. >> >> It would seem that both phones are on the local net via VPN, and should be >> able to talk to each other if they can talk to anyone in the office. (As you >> know.) So look for clues as to real paths to each other. >> >> Cary Fitch >> > > -- > _ > -- 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 > -- _ -- 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
[asterisk-users] simple sip question (I think)
I'm having a strange problem with a sip client and 2 asterisk servers connected together with a sip trunk. Here's a rough layout sip_client -- Asterisk A -[sip trunk] -- Asterisk B when the sip client tries to dial an extension on Asterisk B, Asterisk A sends the invite to B using "sip_cli...@[ip address of asterisk A]" rather than the username A uses to talk to B. Of course B refuses the invite because sip_client registers to asterisk A, not to asterisk B. If a non-sip device connected to A phones the same extension on B, everything works fine. I have insecure=invite on both A and B. I don't understand why A tries to send the invite with the username its client registers to A with, rather than the username A registers to B with. When the incomming call to A (destined for B) doesn't originate on SIP (unistim in this case), obviously A correctly uses the username it registers at B with, as the call goes through. What am I missing here? Andrew ___ -- 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
Re: [asterisk-users] Problem with Asterisk and SPA-3000
The SPA-3000 is notorious for falsely detecting DTMF tones in regular voice, and when it "thinks" it hears DTMF, it will produce a short real DTMF tone that's only audible to the SIP side of the device, not the PSTN side, or out of band SIP DTMF message (dependent on how you have the device setup). I'm guessing you have your IVR setup so that a single keypress triggers it, and the SPA-3000 thinks it hears that key. I would suggest to not have any single key start your IVR, as the SPA-3000 only ever seems to incorrectly detect and generate one DTMF tone at a time (it never generates a sequence, just one random key's worth of DTMF). Hope this helps, Andrew On Wed, Dec 9, 2009 at 7:35 AM, Ivan Stepaniuk wrote: > Hello everybody, > > I have a very strange issue with a Linksys SPA-3000 (1 fxo + 1 fxs) used > as PSTN gateway to asterisk in a small office. Everything works just > fine, except that sometimes, and it seems that only for long incoming > calls, the IVR menu appears on the middle of the call(like a three way > call, call goes on with prompts playing over the parties). Dialing an > extension at the prompt at that time actually works but disconnects the > original extension (and transfers the PSTN leg to the new extension as > normally). > > At the CLI there is nothing but a new incoming call from the SPA, > exactly as the original call. > > It seems to happen with both asterisk 1.2 and 1.4, I am quite lost, Does > anyone know what could be causing this problem? > > > -- > Iván Stepaniuk > Alba Fotónica S.L. > www.albafotonica.com > > > ___ > -- 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 > ___ -- 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
[asterisk-users] dialplan pattern matching
Hi Is there anyway to add logic to dialplan pattern matching? I would like to match all toll free numbers with one pattern, so 1800, 1877, 1866, 1855, etc. I can't figure out how to do this in dialplan syntax. As a programmer, I want to say 18[00 or 77 or 66 or 55 etc]. Can't figure out if this is even possible with dialplan pattern matching (though I suspect it is somehow). Andrew ___ -- 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
Re: [asterisk-users] Cisco SPA3102 Thoughts & Other Recommendations
Did you try changing the FXO port impedance? Did you try resetting the unit to factory defaults (on the IVR dial 73738# and then 1 to confirm)? Did you by it from ebay and it shipped from hong kong or similar? Did you try the linksys silver one on the same line as the real Sipura one was on? I ask because I have two of the the Linksys silver ones (and they are from ebay from hong kong - supposedly "new in box" they are anything but new in box - I've read that they were warranty returns shipped to China to be destroyed, but instead the chinese attempt to patch them up and sell them again - be careful on ebay). One doesn't ring the FXS port properly as there is something wrong with the boost converter that generates the -60V for ringing (and I've spent at least 4 or 5 hours looking at datasheets of the chips, testing parts with a multimeter, looking at signals with a scope, and even swapping parts between the one that works and the one that doesn't and can't figure out what's wrong - I can hear a "tick tick tick..." noise that isn't the relay, and the ethernet led dims with the clicks, like something is breaking down and shorting at the high voltage), but both are currently in service on two different PSTN lines and I have no echo problems with them. Andrew On Wed, Nov 4, 2009 at 5:07 PM, Joseph wrote: > On 11/04/09 16:01, Andrew Hakman wrote: >>On Wed, Nov 4, 2009 at 1:44 PM, Joseph wrote: >>> On 11/04/09 15:20, Adam Tauno Williams wrote: >>> I have two of these and experience a lot of echo on PSTN line (FXS line >>> works OK). >>> The echo is almost impossible to get rid of, so test it first before you >>> buy this unit; Google for "SPA3102 echo" and you will notice I'm not the >>> only one. >> >>Is the SPA-3102 different than the SPA-3000 in this regard? >> >>I have 2 SPA-3000s and have no problems with echo on the FXO port. I >>was thinking about buying some SPA-3102s as the 3000s are hard to find >>now, so if the 3102 has the issue and the 3000 doesn't, I would sure >>like to know about it. I won't buy any 3102s if that's the case. >> >>Andrew > > I have two original Sipura SPA-3000 (dark green boxes) and they work like a > charm, no echo fax is going through IN/OUT every time) they work almost > perfect. > However, I bought a used one SPA-3000 (silver case) it was made after Linksys > took over from Sipura and this unit is a piece of junk, it is even worse then > SPA-3102, the PSTN line has so much echo it is impossible to conduct the > conversation and trying to send or receive a fax fails every time; I've tried > different firmware without any success, stay away from Linksys units. \ > I think they couldn't fix it and sold the unit to Cisco :-/ (my guessing, I > don't know what is the story), but they don't work for me. > The only port I was able to use is FXS. > > So, look for luck somewhere else. > > -- > Joseph > > ___ > -- 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 > ___ -- 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
Re: [asterisk-users] Cisco SPA3102 Thoughts & Other Recommendations
On Wed, Nov 4, 2009 at 1:44 PM, Joseph wrote: > On 11/04/09 15:20, Adam Tauno Williams wrote: > I have two of these and experience a lot of echo on PSTN line (FXS line works > OK). > The echo is almost impossible to get rid of, so test it first before you buy > this unit; Google for "SPA3102 echo" and you will notice I'm not the only one. Is the SPA-3102 different than the SPA-3000 in this regard? I have 2 SPA-3000s and have no problems with echo on the FXO port. I was thinking about buying some SPA-3102s as the 3000s are hard to find now, so if the 3102 has the issue and the 3000 doesn't, I would sure like to know about it. I won't buy any 3102s if that's the case. Andrew ___ -- 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
Re: [asterisk-users] Concurrent calls including mysql taking lot of time for execution
Hey now, I'm a "newschool" programmer and I use vim (and vi, when necessary). Andrew On Wed, Oct 21, 2009 at 8:02 PM, Jeff LaCoursiere wrote: > >> Steve Edwards wrote: >> >>> Since I'm an "old-school" C programmer, I use emacs as my editor. I fire >>> up gdb (the GNU C (amongst other languages) debugger) in a window, give it >>> a command like "b main; r >> through my program line by line, examining and changing variables at will. >>> > > Bah. If you were really old school you would use vi. [ducking!] :) > > j > > ___ > -- 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 > ___ -- 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
Re: [asterisk-users] IAX problem through intermediate asterisk box
I initially had no trunking anywhere, and had the same behavior. I
thought trunking would help, but I can't figure out why the /dev/dahdi
device doesn't get created on C. The dahdi tools / modules don't seem
to have much error / debugging info available, or if they do, I sure
can't find it anywhere obvious.
Andrew
On Thu, Mar 26, 2009 at 11:39 PM, Brandon B. wrote:
> Here's my troubleshooting help -- since the problem sounds like a timing
> issue and part of the call is being trunked, then fix your timing problem,
> or remove the trunking from A and B then see if the problem goes away.
>
> On Thu, Mar 26, 2009 at 10:50 PM, Andrew Hakman
> wrote:
>>
>> So no one else has a problem routing IAX traffic through an
>> intermediate Asterisk server? Does anyone else use Asterisk in such a
>> configuration?
>>
>> On Thu, Mar 26, 2009 at 2:45 AM, Andrew Hakman
>> wrote:
>> > I'm having a problem with IAX running through an intermediate asterisk
>> > box. Perhaps a small diagram will explain the situation better:
>> >
>> > *A --- [cloud (public internet)] --- *B [cloud
>> > (private network)]--- *C
>> >
>> > Asterisk server's A, B, and C, are all connected together with IAX
>> > All asterisk servers are 1.6.0.6
>> > Server A and B are geographically close, but connected over the public
>> > internet.
>> > Server B and C are geographically far, but connected over a private
>> > network.
>> > (the latency between A and B, and B and C are roughly equal)
>> >
>> > Each server has at least 1 phone hanging off of it, with A and C
>> > having most of the phones (B only has a couple).
>> > A's extensions are all 1XXX, B's are 2XXX, and C's are 3XXX
>> >
>> > Phoning from A to B (or vice versa) works well, as does phoning from B
>> > to C (and vice versa). Calls can be placed for an indefinite amount of
>> > time and everything works great.
>> >
>> > The problem arises when phoning from A through B to C (or vice versa).
>> > For the first small amount of time (which can vary on a call to call
>> > basis, and lasts from 0 seconds to 3 minutes or so) everything is
>> > fine. After this, the audio in both directions gets garbled, and
>> > starts arriving in spurts. Once this happens, it continues forever.
>> > The audio never returns to normal no matter how long you wait.
>> >
>> > A to B uses IAX with trunking. B to C is not using trunking
>> > (dahdi_dummy is not working well on C for some reason - the module
>> > loads, but no /dev/dahdi is ever created). The same behavior happens
>> > when A to B is not using trunking either.
>> >
>> > Usually only 1 call is being placed at a time. An interesting thing
>> > happens when 2 testcalls are in progress at the same time though. If
>> > there's a call from A to B, and a call from A to C is made, once the
>> > call from A to C becomes garbled, so does the A to B call. When the A
>> > to C call is ended, the A to B call clears up. Ending the A to B call
>> > first does not improve the A to C call.
>> >
>> > The dialplans are setup so each server passes all non-local extensions
>> > to it's neighbor.
>> >
>> > Hence, for A, the relevant part of the dialplan is
>> >
>> > exten => _2XXX,1,Verbose(1|Extension 2xxx)
>> > exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
>> > exten => _2XXX,n,Hangup()
>> >
>> > exten => _3XXX,1,Verbose(1|Extension 3xxx)
>> > exten => _3xxx,n,Dial(IAX2/asterisk_B/${EXTEN})
>> > exten => _3xxx,n,Hangup()
>> >
>> > For B:
>> >
>> > exten => _1XXX,1,NoOp()
>> > exten => _1XXX,n,Dial(IAX2/asterisk_A/${EXTEN})
>> > exten => _1XXX,n,Hangup()
>> >
>> > exten => _3xxx,1,NoOp()
>> > exten => _3xxx,n,Dial(IAX2/asterisk_C/${EXTEN})
>> > exten => _3xxx,n,Hangup()
>> >
>> >
>> > For C:
>> > exten => _2XXX,1,Verbose(1|Extension 2xxx)
>> > exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
>> > exten => _2XXX,n,Hangup()
>> >
>> > exten => _1XXX,1,Verbose(1|Extension 1xxx)
>> > exten => _1XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
>> > exten => _1XXX,n,Hangup()
>> >
>> > Is this the proper way to set such a configuration up? Is there a
>> > better way to call from A through B to C that would work better?
&
Re: [asterisk-users] IAX problem through intermediate asterisk box
I'll have to get some VPN's setup, but I will give it a try with SIP.
Thanks for the input - you saved me building 2 more asterisk servers
for testing this issue locally (rather than across 3 networks).
Andrew
On Thu, Mar 26, 2009 at 11:12 PM, Steve Totaro
wrote:
> On Fri, Mar 27, 2009 at 12:50 AM, Andrew Hakman
> wrote:
>> So no one else has a problem routing IAX traffic through an
>> intermediate Asterisk server? Does anyone else use Asterisk in such a
>> configuration?
>>
>> On Thu, Mar 26, 2009 at 2:45 AM, Andrew Hakman
>> wrote:
>>> I'm having a problem with IAX running through an intermediate asterisk
>>> box. Perhaps a small diagram will explain the situation better:
>>>
>>> *A --- [cloud (public internet)] --- *B [cloud
>>> (private network)]--- *C
>>>
>>> Asterisk server's A, B, and C, are all connected together with IAX
>>> All asterisk servers are 1.6.0.6
>>> Server A and B are geographically close, but connected over the public
>>> internet.
>>> Server B and C are geographically far, but connected over a private network.
>>> (the latency between A and B, and B and C are roughly equal)
>>>
>>> Each server has at least 1 phone hanging off of it, with A and C
>>> having most of the phones (B only has a couple).
>>> A's extensions are all 1XXX, B's are 2XXX, and C's are 3XXX
>>>
>>> Phoning from A to B (or vice versa) works well, as does phoning from B
>>> to C (and vice versa). Calls can be placed for an indefinite amount of
>>> time and everything works great.
>>>
>>> The problem arises when phoning from A through B to C (or vice versa).
>>> For the first small amount of time (which can vary on a call to call
>>> basis, and lasts from 0 seconds to 3 minutes or so) everything is
>>> fine. After this, the audio in both directions gets garbled, and
>>> starts arriving in spurts. Once this happens, it continues forever.
>>> The audio never returns to normal no matter how long you wait.
>>>
>>> A to B uses IAX with trunking. B to C is not using trunking
>>> (dahdi_dummy is not working well on C for some reason - the module
>>> loads, but no /dev/dahdi is ever created). The same behavior happens
>>> when A to B is not using trunking either.
>>>
>>> Usually only 1 call is being placed at a time. An interesting thing
>>> happens when 2 testcalls are in progress at the same time though. If
>>> there's a call from A to B, and a call from A to C is made, once the
>>> call from A to C becomes garbled, so does the A to B call. When the A
>>> to C call is ended, the A to B call clears up. Ending the A to B call
>>> first does not improve the A to C call.
>>>
>>> The dialplans are setup so each server passes all non-local extensions
>>> to it's neighbor.
>>>
>>> Hence, for A, the relevant part of the dialplan is
>>>
>>> exten => _2XXX,1,Verbose(1|Extension 2xxx)
>>> exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
>>> exten => _2XXX,n,Hangup()
>>>
>>> exten => _3XXX,1,Verbose(1|Extension 3xxx)
>>> exten => _3xxx,n,Dial(IAX2/asterisk_B/${EXTEN})
>>> exten => _3xxx,n,Hangup()
>>>
>>> For B:
>>>
>>> exten => _1XXX,1,NoOp()
>>> exten => _1XXX,n,Dial(IAX2/asterisk_A/${EXTEN})
>>> exten => _1XXX,n,Hangup()
>>>
>>> exten => _3xxx,1,NoOp()
>>> exten => _3xxx,n,Dial(IAX2/asterisk_C/${EXTEN})
>>> exten => _3xxx,n,Hangup()
>>>
>>>
>>> For C:
>>> exten => _2XXX,1,Verbose(1|Extension 2xxx)
>>> exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
>>> exten => _2XXX,n,Hangup()
>>>
>>> exten => _1XXX,1,Verbose(1|Extension 1xxx)
>>> exten => _1XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
>>> exten => _1XXX,n,Hangup()
>>>
>>> Is this the proper way to set such a configuration up? Is there a
>>> better way to call from A through B to C that would work better?
>>> Anyone else experience total audio breakup after a while with a
>>> similar arrangement? Why does it work initially for up to about 3
>>> minutes, then completely fall apart?
>>>
>>> Thanks,
>>> Andrew
>>>
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailin
Re: [asterisk-users] IAX problem through intermediate asterisk box
So no one else has a problem routing IAX traffic through an
intermediate Asterisk server? Does anyone else use Asterisk in such a
configuration?
On Thu, Mar 26, 2009 at 2:45 AM, Andrew Hakman wrote:
> I'm having a problem with IAX running through an intermediate asterisk
> box. Perhaps a small diagram will explain the situation better:
>
> *A --- [cloud (public internet)] --- *B [cloud
> (private network)]--- *C
>
> Asterisk server's A, B, and C, are all connected together with IAX
> All asterisk servers are 1.6.0.6
> Server A and B are geographically close, but connected over the public
> internet.
> Server B and C are geographically far, but connected over a private network.
> (the latency between A and B, and B and C are roughly equal)
>
> Each server has at least 1 phone hanging off of it, with A and C
> having most of the phones (B only has a couple).
> A's extensions are all 1XXX, B's are 2XXX, and C's are 3XXX
>
> Phoning from A to B (or vice versa) works well, as does phoning from B
> to C (and vice versa). Calls can be placed for an indefinite amount of
> time and everything works great.
>
> The problem arises when phoning from A through B to C (or vice versa).
> For the first small amount of time (which can vary on a call to call
> basis, and lasts from 0 seconds to 3 minutes or so) everything is
> fine. After this, the audio in both directions gets garbled, and
> starts arriving in spurts. Once this happens, it continues forever.
> The audio never returns to normal no matter how long you wait.
>
> A to B uses IAX with trunking. B to C is not using trunking
> (dahdi_dummy is not working well on C for some reason - the module
> loads, but no /dev/dahdi is ever created). The same behavior happens
> when A to B is not using trunking either.
>
> Usually only 1 call is being placed at a time. An interesting thing
> happens when 2 testcalls are in progress at the same time though. If
> there's a call from A to B, and a call from A to C is made, once the
> call from A to C becomes garbled, so does the A to B call. When the A
> to C call is ended, the A to B call clears up. Ending the A to B call
> first does not improve the A to C call.
>
> The dialplans are setup so each server passes all non-local extensions
> to it's neighbor.
>
> Hence, for A, the relevant part of the dialplan is
>
> exten => _2XXX,1,Verbose(1|Extension 2xxx)
> exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
> exten => _2XXX,n,Hangup()
>
> exten => _3XXX,1,Verbose(1|Extension 3xxx)
> exten => _3xxx,n,Dial(IAX2/asterisk_B/${EXTEN})
> exten => _3xxx,n,Hangup()
>
> For B:
>
> exten => _1XXX,1,NoOp()
> exten => _1XXX,n,Dial(IAX2/asterisk_A/${EXTEN})
> exten => _1XXX,n,Hangup()
>
> exten => _3xxx,1,NoOp()
> exten => _3xxx,n,Dial(IAX2/asterisk_C/${EXTEN})
> exten => _3xxx,n,Hangup()
>
>
> For C:
> exten => _2XXX,1,Verbose(1|Extension 2xxx)
> exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
> exten => _2XXX,n,Hangup()
>
> exten => _1XXX,1,Verbose(1|Extension 1xxx)
> exten => _1XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
> exten => _1XXX,n,Hangup()
>
> Is this the proper way to set such a configuration up? Is there a
> better way to call from A through B to C that would work better?
> Anyone else experience total audio breakup after a while with a
> similar arrangement? Why does it work initially for up to about 3
> minutes, then completely fall apart?
>
> Thanks,
> Andrew
>
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[asterisk-users] IAX problem through intermediate asterisk box
I'm having a problem with IAX running through an intermediate asterisk
box. Perhaps a small diagram will explain the situation better:
*A --- [cloud (public internet)] --- *B [cloud
(private network)]--- *C
Asterisk server's A, B, and C, are all connected together with IAX
All asterisk servers are 1.6.0.6
Server A and B are geographically close, but connected over the public internet.
Server B and C are geographically far, but connected over a private network.
(the latency between A and B, and B and C are roughly equal)
Each server has at least 1 phone hanging off of it, with A and C
having most of the phones (B only has a couple).
A's extensions are all 1XXX, B's are 2XXX, and C's are 3XXX
Phoning from A to B (or vice versa) works well, as does phoning from B
to C (and vice versa). Calls can be placed for an indefinite amount of
time and everything works great.
The problem arises when phoning from A through B to C (or vice versa).
For the first small amount of time (which can vary on a call to call
basis, and lasts from 0 seconds to 3 minutes or so) everything is
fine. After this, the audio in both directions gets garbled, and
starts arriving in spurts. Once this happens, it continues forever.
The audio never returns to normal no matter how long you wait.
A to B uses IAX with trunking. B to C is not using trunking
(dahdi_dummy is not working well on C for some reason - the module
loads, but no /dev/dahdi is ever created). The same behavior happens
when A to B is not using trunking either.
Usually only 1 call is being placed at a time. An interesting thing
happens when 2 testcalls are in progress at the same time though. If
there's a call from A to B, and a call from A to C is made, once the
call from A to C becomes garbled, so does the A to B call. When the A
to C call is ended, the A to B call clears up. Ending the A to B call
first does not improve the A to C call.
The dialplans are setup so each server passes all non-local extensions
to it's neighbor.
Hence, for A, the relevant part of the dialplan is
exten => _2XXX,1,Verbose(1|Extension 2xxx)
exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
exten => _2XXX,n,Hangup()
exten => _3XXX,1,Verbose(1|Extension 3xxx)
exten => _3xxx,n,Dial(IAX2/asterisk_B/${EXTEN})
exten => _3xxx,n,Hangup()
For B:
exten => _1XXX,1,NoOp()
exten => _1XXX,n,Dial(IAX2/asterisk_A/${EXTEN})
exten => _1XXX,n,Hangup()
exten => _3xxx,1,NoOp()
exten => _3xxx,n,Dial(IAX2/asterisk_C/${EXTEN})
exten => _3xxx,n,Hangup()
For C:
exten => _2XXX,1,Verbose(1|Extension 2xxx)
exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
exten => _2XXX,n,Hangup()
exten => _1XXX,1,Verbose(1|Extension 1xxx)
exten => _1XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
exten => _1XXX,n,Hangup()
Is this the proper way to set such a configuration up? Is there a
better way to call from A through B to C that would work better?
Anyone else experience total audio breakup after a while with a
similar arrangement? Why does it work initially for up to about 3
minutes, then completely fall apart?
Thanks,
Andrew
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