Re: [asterisk-users] understanding what h extension does [ISSUE SOLVED]
On Tuesday 03 April 2007 07:48, Alan Chandler wrote:
> I am trying to make a dialplan that when I dial 90 I can go round a
> whole set of extensions and leave them a short message, hangup and go
> on the next one.
>
> I use the M facility of dial, with something like this
>
> [messages]
>
> exten => 90,n(calcnextchan),Set(DIALCHAN=...)
> exten => 90,n,Dial(${DIALCHAN},30,M(domessage))
> exten => 90,n,Goto(calcnextchan)
>
> [macro-domessage]
>
> exten => s,1,Playback(message)
> exten => s,n,Set(MACRO_RESULT=CONTINUE)
>
> [There is actually more logic to check for busy dial channels and
> retry them later]
>
> This seems to work fine until one of the callees hangs up before the
> message is played. at which point my call is terminated.
>
OK, its a logic problem. If the caller hangs up before playback is
complete MACRO_RESULT has not been set, so the call is bridged and then
hung up. If I set MACRO_RESULT as the first action of the call macro,
then any interruption from the far end hanging up means that the
dialplan just continues without the call having been bridged.
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[asterisk-users] understanding what h extension does
I am trying to make a dialplan that when I dial 90 I can go round a
whole set of extensions and leave them a short message, hangup and go
on the next one.
I use the M facility of dial, with something like this
[messages]
exten => 90,n(calcnextchan),Set(DIALCHAN=...)
exten => 90,n,Dial(${DIALCHAN},30,M(domessage))
exten => 90,n,Goto(calcnextchan)
[macro-domessage]
exten => s,1,Playback(message)
exten => s,n,Set(MACRO_RESULT=CONTINUE)
[There is actually more logic to check for busy dial channels and retry
them later]
This seems to work fine until one of the callees hangs up before the
message is played. at which point my call is terminated.
I was wondering if I should user the h extension here to pickup the
hungup call from the callee and continue. However I am worried that I
might end up looping if I hang up my end of the call - since I want it
to stop if I do that.
I can't find a definitive explanation of what causes the h extension to
be called. Can someone explain what what happen if I added something
like
exten => h,1,Goto(90,calcnextchan)
to the [messages] context
--
Alan Chandler
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[asterisk-users] Quick question about time
Does the GotoIfTime application work in the local time of the server, or UTC? -- Alan Chandler http://www.chandlerfamily.org.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Understanding the dial flags
I am trying to make a system where a conference user can invite others
to join. I am running the 1.2 version of asterisk, so can't use the
example on voip-info.org.
With use of the X flag on a meetme conference to exit with a single
digit, I can get people to join me in a conference with
exten =>
_XXX,1,Dial(${THEIR_EXTEN},,dG(conference-context^${CALLERID}^1))
where the conference-context has a something like this
[conference-context]
exten => _XXX,1,MeetMe(${EXTEN},XMsa)
exten => _XXX,2,MeetMe(${EXTEN},Ms)
The problem with that approach is that you never get to talk to the
called party and ask whether they want to join the conference or not.
So I thought I would take another approach and try change the Dial
statement
exten => _XXX,1,Set(GOTO_ON_BLINDXFER=conference^${CALLERID}^2)
exten => _XXX,2,Dial(${THEIR_EXTEN},,dTg)
exten => _XXX,3,GoTo(conference-context,${EXTEN},1)
Now, when I blind xfer the user he goes straight into the conference.
Unfortunately, the g flag - which should mean I carry on down the dial
plan doesn't appear to work in this case. When he hangs up, I do go
back to the conference, but when he is transfered and goes to the
conference, I seem to end in limbo.
I even tryed to add an h extension to the same context as the Dial is
in, but that didn't seem to help.
Is there a way out of this?
--
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Re: [asterisk-users] [SOLVED]Linksys SPA 3102 causing me problems
On Thursday 29 March 2007 23:26, Alan Chandler wrote: > I have a linksys SPA 3102 with a DECT phone connected into its > Telephone port. > > It has been working, but something I've done (and I don't know what) > means that now everytime asterisk tries to dial it, it says it is > busy. In the end I did a factory reset and then reloaded all my parameters, and now it works great. -- Alan Chandler http://www.chandlerfamily.org.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA 3102 causing me problems
On Friday 30 March 2007 08:11, Gergo Csibra wrote: > Friday, March 30, 2007, 5:02:08 AM, Matt wrote: > > On 3/29/07, Alan Chandler <[EMAIL PROTECTED]> wrote: > >> I have a linksys SPA 3102 with a DECT phone connected into its > >> Telephone port. > >> > >> It has been working, but something I've done (and I don't know > >> what) means that now everytime asterisk tries to dial it, it says > >> it is busy. > >> > >> I can make calls from it through asterisk > >> > >> I am at a complete loss to know what to try next to fix it. Any > >> ideas? > > > > I dont know if you have done this but run a sip show peers and make > > sure that its registered with asterisk. Sounds like it is not > > registering with asterisk which would allow you to call out but > > when it tries to call you it dosent have an ip to contact you at. > > Wehh... > He activated the DND function of Linksys. It can be activate with *78 > and deactivate with *79. No - thats not it (I don't think). First off DND Setting: is set to no in the User1 page. Secondly I did the *79 to deactivate it if it was set, and then tried again. Still no joy -- Alan Chandler http://www.chandlerfamily.org.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA 3102 causing me problems
On Friday 30 March 2007 04:02, Matt Putnam wrote: > I dont know if you have done this but run a sip show peers and make > sure that its registered with asterisk. Sounds like it is not > registering with asterisk which would allow you to call out but when > it tries to call you it dosent have an ip to contact you at. Just a thought - does asterisk have to register with it? -- Alan Chandler http://www.chandlerfamily.org.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA 3102 causing me problems
On Friday 30 March 2007 04:02, Matt Putnam wrote: > I dont know if you have done this but run a sip show peers and make > sure that its registered with asterisk. Sounds like it is not > registering with asterisk which would allow you to call out but when > it tries to call you it dosent have an ip to contact you at. ship show peers shows it thus roo*CLI> sip show peers Name/username HostDyn Nat ACL Port Status chandler/chandler 192.168.0.10 D 5060 OK (6 ms) 1 sip peers [1 online , 0 offline] sip show registry shows nothing. With sip debug on I get the following as I start to make the call 13 headers, 12 lines Reliably Transmitting (no NAT) to 192.168.0.10:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK1ff726a0;rport From: "Alan" ;tag=as3e6d82c9 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 30 Mar 2007 06:07:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 263 v=0 o=root 29004 29004 IN IP4 192.168.0.20 s=session c=IN IP4 192.168.0.20 t=0 0 m=audio 19854 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- roo*CLI> <-- SIP read from 192.168.0.10:5060: SIP/2.0 100 Trying To: From: "Alan" ;tag=as3e6d82c9 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK1ff726a0 Server: roo.home Content-Length: 0 --- (8 headers 0 lines) --- roo*CLI> <-- SIP read from 192.168.0.10:5060: SIP/2.0 486 Busy Here To: ;tag=cf4213264eacc5ei0 From: "Alan" ;tag=as3e6d82c9 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK1ff726a0 Server: roo.home Content-Length: 0 --- (8 headers 0 lines) --- Transmitting (no NAT) to 192.168.0.10:5060: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK1ff726a0;rport From: "Alan" ;tag=as3e6d82c9 To: ;tag=cf4213264eacc5ei0 Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- dial status is BUSY Destroying call '[EMAIL PROTECTED]' > > On 3/29/07, Alan Chandler <[EMAIL PROTECTED]> wrote: > > I have a linksys SPA 3102 with a DECT phone connected into its > > Telephone port. > > > > It has been working, but something I've done (and I don't know > > what) means that now everytime asterisk tries to dial it, it says > > it is busy. > > > > I can make calls from it through asterisk > > > > I am at a complete loss to know what to try next to fix it. Any > > ideas? > > > > > > -- > > Alan Chandler > > http://www.chandlerfamily.org.uk > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users -- Alan Chandler http://www.chandlerfamily.org.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys SPA 3102 causing me problems
I have a linksys SPA 3102 with a DECT phone connected into its Telephone port. It has been working, but something I've done (and I don't know what) means that now everytime asterisk tries to dial it, it says it is busy. I can make calls from it through asterisk I am at a complete loss to know what to try next to fix it. Any ideas? -- Alan Chandler http://www.chandlerfamily.org.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfering not working - how to debug?
I cannot seem to get any transfers to work at all. The console show I
have #1 amd #2 set up for Blind and Attended Transfer, but when I hit
these buttons on my handset nothing happens (other than I hear the dtmf
tones on the other end of the line).
roo*CLI> show features
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# #1
Attended Transfer #2
One Touch Monitor *1
Disconnect Call * *0
I am using the tT options in my dial calls (via a macro)
[macro-extension]
exten => s,1,Dial(${ARG1},20,tT)
--
Alan Chandler
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Re: [asterisk-users] Problem with ztdummy
On Monday 26 March 2007 05:27, Tzafrir Cohen wrote: > You have the module zaptel from an older version still loaded . > > /etc/init.d/zaptel unload > /etc/init.d/zaptel start That didn't fix it - but a reboot of the system did, so I am now working perfectly - thanks -- Alan Chandler http://www.chandlerfamily.org.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with ztdummy
On Sunday 25 March 2007 19:33, Tzafrir Cohen wrote: > On Sun, Mar 25, 2007 at 07:14:22PM +0100, Alan Chandler wrote: > > On Sunday 25 March 2007 11:07, Tzafrir Cohen wrote: > > > On Sun, Mar 25, 2007 at 07:43:19AM +0100, Alan Chandler wrote: > > > > ... > > > > > aptitude search zaptel | grep ^i > > > > Ah - I had no idea I needed this zaptel package. the Asterisk > > package doesn't even have it as a suggests!. > > > > Actually, this looks more complex than I thought from the readme, > > needs some work - so I'll try that and come back > > Would you mind answering that simple question? before aptitude search zaptel | grep ^i gave no output. I have now installed zaptel and zaptel-source and used m-a to build the modules. Now I get i zaptel - zapata telephony utilities i zaptel-modules-2.6.18-4-686 - zaptel modules for Linux (kernel 2.6.18-4- i zaptel-source - Zapata telephony interface (source code fo > > The package zaptel include ztcfg as well as the zaptel init.d script. > It should not be strictly needed for ztdummy, but with it ztdummy > will be loaded automatically. If I run /etc/init.d/zaptel start I get the following error [EMAIL PROTECTED]:/etc/init.d# /etc/init.d/zaptel start Zaptel telephony kernel driver: FATAL: Error inserting ztdummy (/lib/modules/2.6.18-4-686/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) zaptel. and looking at the tail of syslog I get Mar 25 22:13:22 roo kernel: ztdummy: disagrees about version of symbol zt_receive Mar 25 22:13:22 roo kernel: ztdummy: Unknown symbol zt_receive Mar 25 22:13:22 roo kernel: ztdummy: disagrees about version of symbol zt_transmit Mar 25 22:13:22 roo kernel: ztdummy: Unknown symbol zt_transmit Mar 25 22:13:22 roo kernel: ztdummy: disagrees about version of symbol zt_unregister Mar 25 22:13:22 roo kernel: ztdummy: Unknown symbol zt_unregister Mar 25 22:13:22 roo kernel: ztdummy: disagrees about version of symbol zt_register Mar 25 22:13:22 roo kernel: ztdummy: Unknown symbol zt_register Mar 25 22:13:22 roo kernel: Registered tone zone 4 (United Kingdom) but lsmod | grep 'zaptel' gives zaptel180420 0 crc_ccitt 2240 1 zaptel ztdummy is NOT loaded. > > I still can't explain the error you got. So in addition to the input > from the command above: what version of Debian it is, exactly? Debian Etch - updated via aptitude as of today. -- Alan Chandler http://www.chandlerfamily.org.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with ztdummy
On Sunday 25 March 2007 11:07, Tzafrir Cohen wrote: > On Sun, Mar 25, 2007 at 07:43:19AM +0100, Alan Chandler wrote: ... > > aptitude search zaptel | grep ^i Ah - I had no idea I needed this zaptel package. the Asterisk package doesn't even have it as a suggests!. Actually, this looks more complex than I thought from the readme, needs some work - so I'll try that and come back -- Alan Chandler http://www.chandlerfamily.org.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with ztdummy
Although I have a Debian system with prebuilt asterisk package available, I was finding it crashed when I tried to use MeetMe. So I have built asterisk from scratch. However the first thing I try and do is install the ztdummy module with modprobe ztdummy but it always fails to load with FATAL: Error inserting ztdummy (/lib/modules/2.6.18-4-686/misc/ztdummy.ko): Device or resource busy Can someone give me any clues as to what is wrong -- Alan Chandler http://www.chandlerfamily.org.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Debian Asterisk and MeetMe
I am trying to set up a simple conference call capability with asterisk My meetme.conf [general] [rooms] conf => 61 conf => 62 conf => 63 conf => 63 My extensions.conf exten => 60,1,Answer() exten => 60,2,MeetMe(,EMxp) When I enter extension 60 I enter a conference - I get repeated "you are entering conference 6 1 that is not a valid conference number you are entering conference 6 1" etc CPU load rises to near 100% from mpeg123 and asterisk has exited. This seems to be the same if I take the M of the options (ie no music on hold). It appears to be something related to the the zap device Mar 23 15:55:49 WARNING[19849]: chan_zap.c:913 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Mar 23 15:55:49 ERROR[19849]: chan_zap.c:7518 chandup: Unable to dup channel: No such file or directory Mar 23 15:55:49 WARNING[19849]: app_meetme.c:465 build_conf: Unable to open pseudo channel - trying device Mar 23 15:55:49 WARNING[19849]: app_meetme.c:468 build_conf: Unable to open pseudo device I presume, being the Debian binary package, that I don't have the ztdummy driver installed. On the otherhand I read in viop-info.org that that wasn't necessary with Linux 2.6 (I am running 2.6.18). Where do I go from here? -- Alan Chandler http://www.chandlerfamily.org.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voice quality issues
Being new to Asterisk, I have set up a little test rig at home. Asterisk itself is the Debian Etch package, running on a Celeron 1.7G machine I have two clients on a 100Mb LAN, a Windows XP machine with an Athlon XP2200+ processor, and a Linux Core Duo 6300 machine. On the windows XP machine I run idefisk. On the linux machine I can run either idefisk or kiax. Both of these are obviously IAX clients. When I connect the two of them together, I am getting quite severe sound quality problems in the direction of the linux to windows machine. The best I can describe it is the sound coming out the windows machine sounds like a Darlek (very tinny sound with high pitched echo sounds). I think it represents an over compression problem. The sound coming the other way (windows to linux) is fine. I was worried about my raw capture capability on linux, so I just tried to record and playback directly on the local machine - but the quality here is good. It also appears that the main media stream is not going through the Asterisk server as I have an iptables firewall that is counting iax packets hitting the machine and it is small (in the few tens). How can I debug whats wrong here so I can try and correct it. -- Alan Chandler http://www.chandlerfamily.org.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tesco Internet Phone
On Thursday 01 March 2007 20:33, bails wrote: > plug it in a linux box and tell us what it is please, > generic-usb-audio or what? > > Bails > > Julian Lyndon-Smith wrote: > > Yeah, that's where firefly comes from, doesn't it. > > > > I've got the base station plugged in, and the handset connected to > > it, but it always says "pc unavailable". > > > > My system (xp) "sees" a "usb phone" for speakers and microphone, > > but I can't get it to work. > Did this go any further. I would be interested in this. Tesco also do a unit which plugs into the ethernet into which you can plug a standard analogue phone. Looking at the Freshtel web site implies it is an IAX device, but wonder what controls which peer it connects to. Does anybody know if I could use it to connect to my own Asterisk server? -- Alan Chandler http://www.chandlerfamily.org.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Planning Help
On Wednesday 28 February 2007 21:08, mail-lists wrote: > > a) to what extent Asterisk can manage everything necessary to allow > > machines A and B to communicate if they were SIP phones. Is it > > possible to go for a setup with the firewalls/NAT devices as shown > > If the asterisk machine isn't NATed you shouldn't have a problem at > all. If you're using SIP clients just make sure nat=yes > is set in each of the client definitions in sip.conf > > > b) if I go with IAX softphones, does communication between A and B > > have to go through S, or can Asterisk "hand-off" the IAX > > conversation so that A and B talk directly. > > I'm not sure in this case since both clients are going to be NATed. > I'm pretty sure that this wouldn't work with SIP clients. Now you have confused me. In the answer to a) you say that for each SIP client I say nat=yes and it will work, yet here you say this wouldn't work if both clients are going to be SIP. > Since IAX has less problems with NAT traversal it might work fine - > try setting canreinvite=yes in your iax.conf and monitor > rtp traffic at the asterisk CLI You have confused me again. I thought the point of IAX is that there isn't any separate RTP traffic. > > > c) the example documentation shows seperate entries in iax.conf for > > incoming and outgoing calls. In my case (assuming IAX softphones) > > would I just have entries for A and B of type friend? > > > > Can someone give me some advice about how to proceed. > > type=friend works for me... I am not sure where all that leaves me. Should I use SIP everywhere or IAX everywhere -- Alan Chandler http://www.chandlerfamily.org.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Planning Help
On Wednesday 28 February 2007 21:26, Andrew Kohlsmith wrote: > On Wednesday 28 February 2007 3:45 pm, Alan Chandler wrote: > > I am trying to setup an arrangement whereby clients on machines A, > > B, C and D can talk to each other on Softphones. A,B,C are are all > > Windows XP machines, machines D and S are linux. This has to > > include A talking to B and ultimately conference calls with > > potentially all parties. > > Personally I make my Asterisk box the firewall. It eliminates all > NAT troubles. :-) Yes thats what I meant. My box S is the firewall and * will run on it. BUT, both A and B will have NAT firewall/routers outside of them AND somehow C and D will need to go through the S (does the traffic go round the outside of * or through the middle of it? > > If that's not your style, I'd use IAX over SIP, as it only requires a > port-forward to D on D's NAT box. SIP you may be able to get work > with port forwarding 5060 and 1-2 (all udp) over to D, but I am not sure I am following. Why is D different from C? if I port forward everything to D how does C get into the conversation > I'm not sure... Naturally, nat=yes and canreinvite=no should be set > all around. Why? and doesn't the canreinvite=no mean all the traffic from A to B goes through S, something I would prefer to avoid. -- Alan Chandler http://www.chandlerfamily.org.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie Planning Help
Excuse the ASCII diagramme - you will need a fixed width font to understand it. -- --- --- - | A | ==> | NAT | === === | NAT | <== | B | -- ---| |--- - --- | The Internet | --- | WAN interface (82.44.22.127) - | S (NAT) | - | LAN interface (192.168.0.20) = | 192.168.0.0/24 range | -- --- | C | | D | -- --- I am at home on machine D (and with wife on machine C), with some family at machines A and B. I am trying to setup an arrangement whereby clients on machines A, B, C and D can talk to each other on Softphones. A,B,C are are all Windows XP machines, machines D and S are linux. This has to include A talking to B and ultimately conference calls with potentially all parties. Machine S is my firewall/router providing NAT services to clients C and D (based soley on my own IPTABLES script) but is ALSO the machine I plan to put Asterisk on (it can therefore bind to two interfaces, with separate configurations for each if I so desire). If appropriate, I could install a STUN server on S. I would prefer if media traffic between A and B avoids using my WAN interface pipe but if that is unavoidable, so be it. I could use SIP or IAX softphones in this setup as long as it is no more complicated that telling A and B what to download and giving them simple setup instructions. They could probably adjust their NAT routers to forward particular ports to them, but its not certain (A shares a flat with others). I have a slight preference for SIP as it means I could potentially replace machines A,B and C with hardware devices in the future. I have been round and round in circles reading the documentation but I am not sure I understand a) to what extent Asterisk can manage everything necessary to allow machines A and B to communicate if they were SIP phones. Is it possible to go for a setup with the firewalls/NAT devices as shown b) if I go with IAX softphones, does communication between A and B have to go through S, or can Asterisk "hand-off" the IAX conversation so that A and B talk directly. c) the example documentation shows seperate entries in iax.conf for incoming and outgoing calls. In my case (assuming IAX softphones) would I just have entries for A and B of type friend? Can someone give me some advice about how to proceed. Thanks -- Alan Chandler http://www.chandlerfamily.org.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie would like some planning advice.
My wife and daughter, and to lesser extent myself and my daughters boyfriend would like a communications system which allowed us to talk to each other, both on a one on one basis, but also occassionally in conference. My wife and I live in a house with an internal LAN with each of us with a desktop machine (hers in Windows XP, mine runs Linux) and a Linux server acting as firewall and NAT router to a broadband connection. This server also runs a bunch of other services (web server, mail server etc). My daughter is away at University but has a broadband connection running windows XP, her boyfried lives at third location also with broadband running windows XP. My wife and daughter would both like to be able to talk whilst walking around the house. My daughter has tried a bluetooth headset connected to a dongle on her computer but found the delays unacceptable. Ideally a DECT handset, which communicates with a basestation that either plugs into a LAN, or one of the computers would seem good and indeed the Philips VOIP1211 would seem to possibly fit the bill and be in a good price range. Unfortunately, all I can find out about its interface capability is that it is "Skype compatible". The same is true of most other handsets that I can find - they say they are Skype compatible without giving any more detail. I was thinking of setting up Asterisk on my Linux Server and providing a limited service to my family as a PBX. But this will only be cost effective if I do not have to make more than a few pounds investment in handsets. With that background, a few questions. 1.) If I keep everything at the SIP/RTP level, can I operate Asterisk on the server along with everything else. Its a 1.7G Celeron, and the loading from the other services is around 5% to 10%. 2.) Can I make these so called "Skype Compatible" handsets work with Asterisk, or failing that are there any recommendations for alternatives which do work, but have the wireless capability described and don't cost a lot. 3.) Can Asterisk manage the NAT traversal that will be necessary for communication from handsets on the WAN and handsets on the LAN. -- Alan Chandler http://www.chandlerfamily.org.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
