Re: [Asterisk-Users] stop monitor on transfer
Anton Krall wrote:
Hi John, yes, Im using native transfer. What I do is use Monitor on the
dialplan of the extension that picks up the call coming from PSTN, so after
that, if the extension forward or transfers the call, monitor keeps
recording all thru the end of the call no matter where it is been
transferred to.
Hmmm. This is what I do:
XX,1,NoOp()
XX,2,MixMonitor(${UNIQUEID}.wav)
XX,3,Dial(SIP/201,15,jTt)
..
The call is then SIP transferred by the receptionist, and that's when
the recording ends.
I'll have a look at native transfer and see if that changes things !
jd
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Re: [Asterisk-Users] stop monitor on transfer
Anton Krall wrote: Guys. This idea has been banging my headfor days now and I feel the need to share with you. Imagine this scenario: all calls come in thru a receptionist, asterisk records all incoming calls, the receptionist's work is to transfer the calls to internal people but some of them are bosses and you know how bosses are, they don't want their calls to be recorded, so, I have been trying to figure a way on how to stop monitoring / recoring calls once they are transferred to a bosses extension while othe transferd to other people stay on record mode. Anton, hi; I've got exactly the opposite problem. I *want* to record the call after the transfer, but (using MixMonitor and SIP transfers on Snom handsets) the recording terminates with the transfer. Are you using Asterisk native transfer ? jd ___ --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] MixMonitor and transferred calls
Hi; I'm trying to record all inbound and outbound calls at a site, and I have a problem with inbound calls that are transferred by a receptionist using Snom's handset buttons (i.e. SIP transfer rather than using the key sequences defined in features.conf). The first leg of the call is recorded fine. There is, however, no recording after the transfer. Am I correct in thinking that I'll have to use Asterisk native transfer for this to work ? jd ___ --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] answer delay
FaberK wrote: Hi guys, maybe youìve got the answer...! When a caller(not internal, but from PSTN) call *, I need to let him hear a message, before * answer and the bill start running. If is not clear, just let me know. caller-telco(telco bill to the caller as soon as * answer)-asterisk Alas, most (if not all) telcos object to you transmitting voice over their circuits before they've started to charge you for the call. I don't think this is possible to implement from the Asterisk end of things. jd ___ --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] answer delay
Andrew Kohlsmith wrote: On Monday 20 March 2006 11:46, John Daragon wrote: Alas, most (if not all) telcos object to you transmitting voice over their circuits before they've started to charge you for the call. Incorrect. I do this all the time with a PRI. You can't do this with POTS. Simply don't Answer() until you're ready to bill. You can send audio but you cannot hear them until you answer the call. Hell, you learn something new every short period of time. I have to go try this out... jd ___ --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] ISDN BRI and UK Premium Rate Numbers
Faris Raouf wrote: Can anyone help point me in the right direction please? I'm based in the UK and I want to start using a Premium Rate number with Asterisk - I think the equivalent in the US would be a 900 number. Effectively the caller pays much more to call such a number than a normal national or local call. The problem with these is that I don't want Asterisk to actually signal to the telephone network that the call has been answered until someone really does answer it, otherwise the caller will be paying a premium rate just to listen to an Asterisk-generated ring tone until someone answers the call. This is pretty standard Asterisk behaviour exten = whatever,1,NoOp exten = whatever,2,Dial(SIP/nSIP/n+1SIP/n+2) exten = whatever,3,Hangup The incoming ISDN call will ring the specified SIP phones, and will not be answered until one of them picks up. Snip Ideally I'd also like the caller and the person answering the call to hear a recorded message saying that calls to this number cost X per minute ... blah blah, this message being triggered only when someone answers the call. This will warn the caller *and* the person answering that this is a premium-rate call. The person answering the call will know to speak after this message has been played. But that's just an ideal situation. Right now I'm more concerned about how to stop Asterisk answering until someone is available to take the call. H ... sorry, no idea how to do this bit - I believe it's a requirement that's been addressed before by implementing a MeetMe conference, but my recollection is hazy... ___ --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] AVM C2 chan_capi-cm-0.6.3 Error on Dial
I'm getting a strange error on one of the two controllers on an AVM C2
card under chan_capi-cm-0.6.3.
I have two ISDN controllers defined, both in the same group, both
connections are UK ISDN2e Point to Point:
On the third outbound call (both of the first two calls are handled by
the second controller ISDN2,) I get this error :
chan_capi.c conf_error 0x2001 PLCI=0x301 Command=CONNECT_B3_CONF,0x8487
Does anyone have any idea what's going on here ? BT tell me there's no
problem they can see with the ISDN line involved.
jd
This is the dialstring :
exten = _9.,1,SetCallerPres(allowed)
exten = _9.,2,SetCIDNum(252000)
exten = _9.,3,Dial(CAPI/g1/${EXTEN:1}/b)
exten = _9.,4,Congestion
/etc/capi.conf contains :
c2 c2.bin DSS1 - - - - P2P
c2 c2.bin DSS1 - - - - P2P
/etc/asterisk/capi.conf contains :
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.9
txgain=0.3
; interface sections ...
[ISDN1]
isdnmode=DID
incomingmsn=*
controller=1
group=1
softdtmf=on
relaxdtmf=on
accountcode=
context=capi-in
holdtype=hold
echocancel=yes
echotail=64
bridge=yes
callgroup=1
devices=2
[ISDN2]
isdnmode=DID
incomingmsn=*
controller=2
group=1
softdtmf=on
relaxdtmf=on
accountcode=
context=capi-in
holdtype=hold
echocancel=yes
echotail=64
bridge=yes
callgroup=1
devices=2
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Re: [Asterisk-Users] AVM C2 chan_capi-cm-0.6.4 Error on Dial
Whoops ! sorry - wrong release ...
chan_capi-cm-0.6.4 !
John Daragon wrote:
I'm getting a strange error on one of the two controllers on an AVM C2
card under chan_capi-cm-0.6.3.
I have two ISDN controllers defined, both in the same group, both
connections are UK ISDN2e Point to Point:
On the third outbound call (both of the first two calls are handled by
the second controller ISDN2,) I get this error :
chan_capi.c conf_error 0x2001 PLCI=0x301 Command=CONNECT_B3_CONF,0x8487
Does anyone have any idea what's going on here ? BT tell me there's no
problem they can see with the ISDN line involved.
jd
This is the dialstring :
exten = _9.,1,SetCallerPres(allowed)
exten = _9.,2,SetCIDNum(252000)
exten = _9.,3,Dial(CAPI/g1/${EXTEN:1}/b)
exten = _9.,4,Congestion
/etc/capi.conf contains :
c2c2.bin DSS1 - - - - P2P
c2c2.bin DSS1 - - - - P2P
/etc/asterisk/capi.conf contains :
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.9
txgain=0.3
; interface sections ...
[ISDN1]
isdnmode=DID
incomingmsn=*
controller=1
group=1
softdtmf=on
relaxdtmf=on
accountcode=
context=capi-in
holdtype=hold
echocancel=yes
echotail=64
bridge=yes
callgroup=1
devices=2
[ISDN2]
isdnmode=DID
incomingmsn=*
controller=2
group=1
softdtmf=on
relaxdtmf=on
accountcode=
context=capi-in
holdtype=hold
echocancel=yes
echotail=64
bridge=yes
callgroup=1
devices=2
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Re: [Asterisk-Users] AVM C2 chan_capi-cm-0.6.4 Error on Dial
Armin Schindler wrote: On Wed, 15 Mar 2006, John Daragon wrote: Whoops ! sorry - wrong release ... chan_capi-cm-0.6.4 ! There were problems with 0.6.3 and AVM cards, but should be fixed in 0.6.4. Can you please create a full debug log (set verbose 5; capi debug) for such a case ? Certainly. Would you like it off-list ? jd ___ --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] Idiot's guide to Q.932?
I've been asked to look at a tender for a switch, and one of the capabilities the customer is looking for is support for Q.932. They have a number of exchanges and are looking, in the future, to support things like remote and aggregated operators. Can anyone point me to an idiot's guide to Q.932 capabilities, please ? jd ___ --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] 10minutes to restart [EMAIL PROTECTED] 2.7
Marco Mouta wrote: Hi all, I've bought a TE110P, and received it today. So i decided to install [EMAIL PROTECTED] 2.7 with this card. In the past i had experiencies with X100P (clone card) and it never take me so long to reboot the machine Have you specified an inaccessible DNS nameserver ? That's usually good for multiple waits of exactly 60 seconds. jd ___ --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] Cannot load wcfxo -- Please help!
Phil Freed wrote: I'm afraid that I am at a loss here. I am new to Asterisk, and have successfully set up SIP. But I cannot get my FXS card working, and I'm not sure what else I can try. # modprobe wcfxo /lib/modules/2.4.21-37.0.1.ELsmp/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.21-37.0.1.ELsmp/misc/wcfxo.o: insmod /lib/modules/2.4.21-37.0.1.ELsmp/misc/wcfxo.o failed /lib/modules/2.4.21-37.0.1.ELsmp/misc/wcfxo.o: insmod wcfxo failed I have a Digium quad card (Freshmaker Rev J) with and FXO daughter card (S110M Rev A) in port 3 and an FXS card (X100M Rev C) in port 4. Are these old cards? Could that be a problem? Snip. IIRC, wcfxo is the driver for the X100P card. The 4 port analog card's driver used to be called wcfxs. but that led to the sort of confusion you're experiencing, so it was renamed to wctdm. js ___ --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] Diff between X100M and X100P?
Phil Freed wrote: I have noticed a lot of folks mentioning the x100P, and very few mentioning x100M (which is what I have). Are there important differences between them? The X100P was a PCI card with a single FXO port (actually a WinModem, more or less). The X100M is a daugterboard for the TDM400P card. jd ___ --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] Operator consoles for large systems
Hi; I've been asked to look at a large asterisk system implementation, which would be a candidate for either a large cluster of PCs or a smaller cluster based on Signate's SGI box(es). I've waded through the requirements document, and I think I have more or less all of the requirements covered with the exception of a group of operator consoles (probably 4 of them). There are, as I see it, a couple of issues (and probably ones that I haven't thought through yet, too!). Please bear with me while I think out loud : 1) physical screen real-estate. This means that we can't use any of the static operator consoles out there - there's just no way to represent 1600 or so users on a PC screen, so we'll have to come up with a way of representing only those users an operator is interested in, and doing so in a way that still lets them use a mouse (and/or keyboard) without everything changing underfoot. I'm thinking something like an old air traffic control strips system for calls requiring service, a phone finder to select where a call is going, and a visible LRU cache of places you may want to send a call. 2) aggregating manager data from many clustered asterisk machines. Obviously we will need some sort of proxy system. The Manager interface is, of course, pretty dynamic, and the approach taken so far seems to have been to parse the manager output and build a graphical representation of state information gleaned in real time. Of course, we'd need to keep much more state than we could display, and it might be more sensible for us to have (perhaps) a state engine for each * machine, and aggregator which in turn broadcasts to Operator clients. Assuming we use 2 bits for the state of each entity, we would be able to describe 2000 users in 4kb (512 bytes) so a frequent broadcast would not be out of the question. Of course, we'd lose data about which endpoints were connected to what. How important have people found that to be in real life ? Sorry for the ramble. Any ideas really, really welcome. jd ___ --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 compiling ztdummy on centos 4, 2.6 kernel
Sina, hi; Let's just do a little recap. You've downloaded zaptel-1.2.4 and done the make linux26 make install make config thing on it. If you don't uncomment anything, the builds complete without error and modules are installed in /lib/modules/`uname -r`/extra. You've performed the 2.6 kernel udev configuration : edit /etc/udev/rules.d/50-udev.rules and insert the lines : KERNEL=zapctl,NAME=zap/ctl KERNEL=zapchannel,NAME=zap/channel KERNEL=zaptimer, NAME=zap/timer KERNEL=zappseudo, NAME=zap/pseudo KERNEL=zap[0-9]*, NAME=zap/%n Assuming you're using a user called asterisk... edit /etc/udev/permissions.d/50-udev.permissions and insert : zap/* asterisk:asterisk:660 If running /etc/init.d/zaptel start still fails, then run /etc/init.d/zaptel stop and then sh -x /etc/init.d/zaptel start You should be able to work out what's failing from the output here. If you can't, post the output to the list or email it to me. If, for example, modprobe is failing on ztdummy.ko, then run strace modprobe ztdummy and look at the output. This will identify problems like the modules being in a directory that modprobe isn't looking at, c c. Again, if the cause isn't clear either post the last (say) 20 lines of the strace err... trace her or email them to me. Let's put this one to bed, huh ? jd ___ --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] ATA's ???
[EMAIL PROTECTED] wrote: Hi, I'm currently in the process of building Asterisk for our new office and have hit a snag. We need two internal Analog lines for a modem and fax machine. Am I right in thinking I can use two ATA's, one on each piece of equipment which will then talk to Asterisk and route via our ISDN30? If the above is corrent could you recommend a good model? Yes, that should work fine - I have a fax machine here connected to a Grandstream Handytone ATA-286 which (with recent firmware) performs faultlessly using G.711 aLaw. I'm not sure how well it will support higer speed modems, though... jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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] Simple setup ...
[EMAIL PROTECTED] wrote: Hi, I'm currently looking to run Asterisk in the office to replace an old PBX and would appreciate a little help. We are moving offices and will have 8 digital lines. My questions are: As there are 8 digital lines is this known as PRI? In the UK, that would be called a partial E1. E1 is the generic name for BT's ISDN30 product. Which Digium card would be the best fit? If it *is* a partial E1, then the Digium TE110P (that's the non-echo-cancelling version) is what you're after. Sangoma make a card with similar capabilities, and others (like Eicon) make E1 cards with DSPs on board which handle echo cancellation as well as a lot of ISDN stack housekeeping. Intelligent cards are *much* more expensive than dumb ones like the TE110P. If you have ISDN2e (4 x 2 channel boxes on the wall), then you'll need a 4 port ISDN2e card. Digium don't make an ISDN2e card. Would you recommend looking at the echo cancellation cards? Sorry, I have no experience of the Digium echo cancellers (yet). We are UK based: is caller id supported by Asterisk for the card that you would recommend? All ISDN30 and ISDN2e cards hadle caller id in the UK. BT (or your alternative carrier) will have to enable it. Anytime anyone asks for advice on a small system, someone wades in with try [EMAIL PROTECTED]. I wouldn't recommend that myself. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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] Fritz card technology German *
Chris Earle (CBL) wrote: Hi all, I've been working with * for a long time now, but only with analog FXS/FXO systems. I am venturing towards setting up a box in Germany now and I believe that requires a Fritz card? Do I even have to use the Fritz cards? Why not a Digium card The AVM Fritz card is a single connection (2 x 64 kbps) passive ISDN card. It's well supported by chan_capi, but running more than one of them in a PC requires a driver patch. You can't use a Digium card because Digium doesn't make an ISDN2 card. We have 2 ISDN lines ( -- 6 handsets) so I'm guessing that will require 2 Fritz PCI cards (they have 1 port only). Then there's some sort of channel bank that sends the calls out to the extensions. Does this make any sort of sense? By 2 lines I guess you mean 4 channels ? i.e. 4 simultaneous calls ? If you mean 2 channels, then you only need 1 fritz card. Could someone confirm with me that this is the right direction to go -- ISDN lines, Fritz cards/Asterisk box, Channelbank/telco-box, extension handsets.. On the handset side you could use a couple of TDM4xx cards, or just use SIP phones. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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] sipura ata 3000 UK (BT) CAllerid
Conrad Wood wrote: Hi I wonder whether anyone got the Sipura ata 3000 to decode British Telecoms callerid and pass it to asterisk? The userguide seems to suggest that this is not possible, is that right? Current firmware handles it beautifully. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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] SOLVED: Hung Zap channels connected to old key system
Philip Edelbrock wrote: We've got a Toshiba DK system w/ analog ports that went to a voicemail server. I swapped in an Asterisk box with a Digium 4-port fxo card. It /almost/ worked perfectly. The problem is that Zap channels never hang up. They have to time out. I set up MeetMe, but all Zap channels hung forever. Very annoying. Same thing for FXO-to-FXO bridges. I figured out today why and fixed it. Some proprietary voicemail systems (and probably tie-lines, too) like to use DTMF tones instead of standard ground/loop/kewl whatever signaling. Our key system was programmed to use such DTMF tones instead of the usual analog signaling on those ports. (I think it was program 31 on our Toshiba DK40i) Asterisk of course ignored those, but the other systems used those for line signaling (including our previous 3rd party system). Amusingly, I know now why for years we kept hearing loud DTMF tones when our branch office picked up their phones. Their system, too, was configured to have those analog lines be connected to a voicemail system and not to a FXO port on a T1 CSU. I've just come across a similar problem with a more modern SpliceCom hybrid PBX. We have an * system connected to two analog (FXS) ports via a couple of Sipura SPA3000 ATAs, and we thought the Sipuras were failing to detect call termination. Turns out that the default behaviour of an FXS port on this PBX is *never* to hang up. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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] Nested MySQL Commands
Douglas Garstang wrote: Peter, I assume you mean something like this in extensions.conf: exten = _X.,1,AGI(master-dial-logic.pl) and then there's only one call. All logic would be performed by the perl script. This has many advantages. One disadvantage however is that potentially, there could be 120 simultaneous instances of this script running (one per call). Yes, but if you need it to scale efficiently, each of these could be a very lightweight process. If you used each of these to communicate via RPC or shared memory to a process with a small and configurable pool of database connections (which isn't that difficult), you can build a simple and scalable solution. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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] Nested MySQL Commands
Douglas Garstang wrote: jd but. but Asterisk still fires up a process each time you make an AGI call in the dialplan. You could still have 120 of these lightweight processes running, in _addition_ to the other process. Doesn't sound like it provides much benefit. True. But Linux is *good* at processes. The code is likely to be in RAM (provided you use a compiled language), And most databases are *bad* (relatively) at new connections, and at large numbers of connections. By keeping connection numbers low, keeping them open, and keeping the number of actual database calls to the minimum I'd be pretty confident of a scalable solution. Remember that these processes will spend most of their lives sleeping. Sure, there'll be more linked-list traversal c, but that's what an OS is for. I'm going to be building just such a beast over the next couple of weeks (for a similar sort of application), perhaps I'll do some performance estimation up front and post it. jd -Original Message- From: John Daragon [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 11, 2006 2:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Nested MySQL Commands Douglas Garstang wrote: Peter, I assume you mean something like this in extensions.conf: exten = _X.,1,AGI(master-dial-logic.pl) and then there's only one call. All logic would be performed by the perl script. This has many advantages. One disadvantage however is that potentially, there could be 120 simultaneous instances of this script running (one per call). Yes, but if you need it to scale efficiently, each of these could be a very lightweight process. If you used each of these to communicate via RPC or shared memory to a process with a small and configurable pool of database connections (which isn't that difficult), you can build a simple and scalable solution. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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] Nested MySQL Commands
Douglas Garstang wrote: jd but. but Asterisk still fires up a process each time you make an AGI call in the dialplan. You could still have 120 of these lightweight processes running, in _addition_ to the other process. Doesn't sound like it provides much benefit. Or, you could have the dialplan pass control to app_access_the_database and have that do the data manipulation, populating dial plan variables for processing as usual (or, I guess, performing the routing itself). No process overhead. I'd still be inclined to use a pool of running, connected database processes/threads to handle the actual queries. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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] bayhamsystems.com experience
Michiel van Baak wrote: Hi all, Anyone using their services ? I'm thinking of setting up my servers with their service. But before starting to mess with my extensions.conf I thought let's check the community for their experience. I don't use them from asterisk, but I do use their SMS service from a locally coded application. Responsive, easy to do business with, absolutely no problems at all. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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] UK ISDN2e with DDI?
Patrick Lidstone (Personal E-mail) wrote:
We're about ready to go ahead with a nice 6 line (maybe later
8) ISDN setup with [EMAIL PROTECTED] and the quad Junghanns card.
Before we do, could anyone confirm for me that BT's ISDN2e
lines do actually provide Asterisk with the DDI number? We
need to be able to route incoming calls based on which number
is ringing.
Yes they do. For DDI ranges you'll need to ask BT for a System Access
installation (sometimes known as Point-to-Point) and configure the
Junghanns appropriately.
I'm probably just an old fogey with a programming background, but I find
straight Asterisk *so* much easier to configure than [EMAIL PROTECTED]
When you say ringtones, do you mean sounds like a UK phone when it
rings, or sounds like a UK phone when we ring someone else ?
If it's the latter, you need early b3 connects so that you hear the
tones generated by BT rather than the ones the Grandstream generates.
With CAPI (which I tend to use), you add the b option in the dial
string. I presume bristuff does something similar.
exten = _0.,3,Dial(CAPI/g1/${EXTEN}/b) ; always ask for early B3
jd
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Re: [Asterisk-Users] UK ISDN2e with DDI?
Dogers wrote: Quoting John Daragon [EMAIL PROTECTED]: snip... When you say ringtones, do you mean sounds like a UK phone when it rings, or sounds like a UK phone when we ring someone else ? It does actually sound okay when we ring someone else, but when it rings, it has the long single american style ring. I've come across a few places that claim its built into the Grandstream and I'd have to create and upload a new one.. but I've also found others that say to edit various config files, which has had no effect (indications.conf and zaptel.conf both have the zone as uk.. Theres nowhere else it needs to be set, is there?). Neither zaptel.conf nor indications.conf will change anything on a SIP phone. You can build and download custom ring tones to later Grandstreams; see: http://www.voip-info.org/wiki-Budgetone and http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone for the description of tools and methods. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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] Can Asterisk act as a media gateway?
Ken D'Ambrosio wrote: I've got an account that's looking at doing some cable/VoIP integration. They were wondering if it were possible to set up something like this: PSTN (T1) - Asterisk - (some VoIP protocol, probably SIP) - Siemens soft switch - their product It sure sounds nice in theory, but I've never tried anything like this. Is there any chance it would work? Yep, we've done ISDN2e -- Asterisk - H.323 - Cisco Call Manager Analogue - Sipura SPA-3000 - which worked really well. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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] UK ISDN2e with DDI?
David Cook wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dogers Sent: 07 December 2005 16:24 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UK ISDN2e with DDI? Quoting John Daragon [EMAIL PROTECTED]: Patrick Lidstone (Personal E-mail) wrote: We're about ready to go ahead with a nice 6 line (maybe later 8) ISDN setup with [EMAIL PROTECTED] and the quad Junghanns card. Before we do, could anyone confirm for me that BT's ISDN2e lines do actually provide Asterisk with the DDI number? We need to be able to route incoming calls based on which number is ringing. Yes they do. For DDI ranges you'll need to ask BT for a System Access installation (sometimes known as Point-to-Point) and configure the Junghanns appropriately. I'll have to check that, I guess - find out what they're set as! I'm probably just an old fogey with a programming background, but I find straight Asterisk *so* much easier to configure than [EMAIL PROTECTED] True, I've used bare Asterisk at home for my small get up, but [EMAIL PROTECTED] just does everything we need it to do here at the office (including the nice and pretty call log side of things that AMP provides!) When you say ringtones, do you mean sounds like a UK phone when it rings, or sounds like a UK phone when we ring someone else ? It does actually sound okay when we ring someone else, but when it rings, it has the long single american style ring. I've come across a few places that claim its built into the Grandstream and I'd have to create and upload a new one.. but I've also found others that say to edit various config files, which has had no effect (indications.conf and zaptel.conf both have the zone as uk.. Theres nowhere else it needs to be set, is there?). Andrew Try adding the following to your handset config in sip.conf. This forces the SIP device to get it's ring tones from Asterisk. Worked for us in v1.0.9 with Polycom handsets. progressinband=yes Be careful when ordering an ISDN2e line from BT. By default they come configured as Point-to-Multipoint with any additional numbers as MSNs. Most PBXs are better with ISDN2e Point-to-Point with DDIs, but BT then sting you for a £100 DDI planning fee in addition to the ISDN2e installation. One thing to consider is that DDIs are allocated in contiguous blocks of 10 numbers e.g. 0115 7889100 - 7889109. MSNs however are purposely allocated by BT randomly in what ever quantity you require. Officially you cannot have contiguous MSNs which aren't so good for PBX use. You described this so much better than I did. IIRC, the first 10 MSNs are contiguous for point to multipoint. After that all bets are off. If you want inbound CLI display (CLIP) and/or the ability to specify the outbound number you are presenting as a CLI ( CLOP/COLP depending on who you are talking to) this needs to be specified as well. By default you get neither but both are non-charegable upgrades (in our limited experience). Our recent PTP installations have turned up without CLIP but with COLP. YMMV, it seems... jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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] ISDN 2e Cards
Faris Raouf wrote: Simon Faulkner wrote: I have tried Billion HFC-PCI and Eicon Diva cards for ISDN 2e in the UK but both seem to have drawbacks/advantages. I need to build a new Asterisk box for my tiny business (1 x ISDN2e from BT and 1 x IAX link from Gradwell) Is anyone prepared to go out on a limb and say which card they prefer and why? TIA Simon ___ --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 BT SpeedWay PCI cards are ideal. Work perfectly with chan_capi-cm for ISDN2e and Business Highway here in the UK. They are basically AVM Fritz cards badged by BT. I have a stock of brand new ones if you need, or alternatively they are often advertised on the auction sites (new and used). I'd second that. For a single ISDN2e connection the AVM Fritz card is really hard to beat/ jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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] Meetme option 'b'
Hi;
I've been looking for an arbitrary way of discovering when the last
user has left a Meetme conference...
It occurred to me that I could launch an agi script to keep watch over
the conference and do something when the user count reaches zero... And
of course, I can do that directly from the dialplan.
But I was looking at app_meetme, and the docs say:
* 'b' — run AGI script specified in ${MEETME_AGI_BACKGROUND}
o Default: conf-background.agi (Note: This does not work
with non-Zap channels in the same conference)
I can't see anything in the code to explain this; does anyone understand
why it might be ?
jd
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Re: [Asterisk-Users] Re: Meetme option 'b'
Tony Mountifield wrote:
In article [EMAIL PROTECTED], John Daragon [EMAIL PROTECTED] wrote:
Hi;
I've been looking for an arbitrary way of discovering when the last
user has left a Meetme conference...
It occurred to me that I could launch an agi script to keep watch over
the conference and do something when the user count reaches zero... And
of course, I can do that directly from the dialplan.
But I was looking at app_meetme, and the docs say:
* 'b' — run AGI script specified in ${MEETME_AGI_BACKGROUND}
o Default: conf-background.agi (Note: This does not work
with non-Zap channels in the same conference)
I can't see anything in the code to explain this; does anyone understand
why it might be ?
To explain which part? That it doesn't work with non-Zap channels?
For Zap channels, the mixing is automatically done at the driver level
once MeetMe has told the driver which channels to mix.
For a non-Zap channel, a proxy Zap channel (pseudo) is created to
participate in the driver-level mix. The meetme thread on the channel
then enters a loop to copy audio back and forth between the non-Zap
channel and the proxy pseudo-channel.
When an AGI background script is specified, it runs INSTEAD OF the
copying loop mentioned above. Therefore there is nothing to move the
audio to and from the non-Zap channel.
Hope this helps!
It does, indeed ! Thanks for the succinct explanation.
I owe you a beer.
jd
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Re: [Asterisk-Users] Would DECT cordless phones work with Asterisk and VOIP?
Larry Alkoff wrote: Snip ... Does anyone have any hands-on experience with DECT? We have an old BT DECT phone in the house, connectd to an FXS port on a TDM400 card. We get a burst of white noise (inaudible to the guys at the other end) for about half a second when we pick up, but apart from that it's fine. Range is about 100m in clear air. We have 3 ft thick stone walls and it copes with that very well. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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] Pros and Cons of T1/E1 cards
Hi; We're looking to standardise on a single family of E1 PRI cards. I guess our options are : Digium / Zaptel / libpri Sangoma/ Zaptel / Wanpipe AVM/ CAPI eIcon / CAPI Junghanns / Bristuff Can anyone share any comparative experience of these, please ? Do they differ much in terms of interrupt requirement, CPU load c ? Any info gratefully received. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] Heads up - AVM C2/C4 on AMD 64 bit processors
I thought I'd just document a problem I've been having in case anyone
else comes across it.
I have an AVM-C2 card which I'm using with chan_capi. capiinit fails to
load the driver when run under 2.4 or 2.6 kernels on AMD Sempron
processors with 64 bit cores, and at least some 64 bit Athlons, too.
If I pull out the RAID and put it into a 32 bit Intel based PC,
everything works fine.
jd.
Here's the symptom (C4 is the driver for C2 and C4 cards) :
PCI: Found IRQ 4 for device :00:7.0
c4: PCI BIOS reports AVM-C2 at i/o e400, irq 4 mem 0xe8124000
c4: NO card at e400 error(2)
c4: no AVM-C2 at i/o e400, irq 4 detected, mem 0xe8124000
c4: revision 1.1.2.2
which means that C4_detect() in C4.c has failed in a chunk of code
that is basically resetting the controller.
What it does is to enter a tight loop for ten seconds, looking for a
value of 0x from a port on the card:
c4outmeml(card-mbase+DOORBELL, DBELL_RESET_ARM);
stop = jiffies + HZ*10;
while (c4inmeml(card-mbase+DOORBELL) != 0x) {
if (!time_before(jiffies, stop))
return 2;
c4outmeml(card-membase+DOORBELL, DBELL_ADDR);
mb();
}
The function fails because it still hasn't read 0x from
card-mbase+DOORBELL after 10 seconds.
Interestingly, if I insert a KERN_DEBUG statement to record the content
of card-mbase+DOORBELL, I get 0x, but the comparison still
fails.
I have been able to persuade the driver to load by extending the time at
HZ*10 to HZ*50 and running capiinit start / capiinit stop a few times,
whiich is obviously not ideal as we can't run it in the init.d script
like that... Under 2.4 kernels it loads silently (when it does load)
under 2.6, it produces a really scary kernel error :
c2-e400: C2-card (3.11-06) now active
Debug: sleeping function called from invalid context at
include/asm/semaphore.h:107
in_atomic():1[expected: 0], irqs_disabled():0
..
..
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Re: [Asterisk-Users] TDM0xB vs. SIP for FXO
Chris Bagnall wrote: This is a very interesting thread. Could folks posting their experiences please also post the country their experiences relate to? We've had very good experience with the SPA-3000 in the UK since the last version of the firmware sorted out local impedance settings (taking, IIRC, the settings in the page and actually applying them to the hardware...) This seemed to fix all of the echo issues we'd seen. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Re: Anyone aware of a current Dell server model with 3PCI slots
Tom Hayden wrote: Yeah, with our Dell Poweredge 750, we had all kinds of IRQ conflicts and whatnot. I booted up and in the BIOS I turned off all sorts of devices, including one of the ethernet cards, the USB, serial, etc. After that, things worked much better. Tom, hi; We have a 750 here - 1 x 64 bit PCI-X and a 32 bit PCI slot. It doesn't seem to matter what we disable in BIOS, both PCI slots share an interrupt. We can chose *which* interrupt they share, but they always share one. You haven't found a way around this (BIOS upgrade/whatever) have you ? jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] OT: Suggestions for E1 Service in the UK
Geoff Manning wrote: We are looking to acquire E1 service in Fleet right outside of London. I am in the States so I am not aware of the key players. We currently get ADSL from Eclipse but were interested in a quote for E1. What is a typical E1 line go for nowadays and who can I get it from? Well, the major incumbent is BT. Are you sitting down ? Installation : Per channel 1 year contract 3/5y contract 3/5y+commitment First 15 channels (min 8)GBP 125 GBP 80GBP 0 16-30 (per channel) GBP 30 GBP 15GBP 0 Annual Rental (per channel) GBP 182.32 DDI Non Quota GBP 208.32 DDI Quota jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] SPA3000 as trunk - no caller ID
Kerry Garrison wrote: During a PSTN call the status screen correctly displays the caller ID information. Well, if the SPA-3000 is picking up the CID, and PSTN CID as VOIP CID is set, and the caller ID isn't being passed to Asterisk, it looks as if the SIP INVITE is being passed to Asterisk before the CID has been detected. But you've obviously thought of that - hence the delay... It may be worth firing up ethereal to check that the CID really isn't in the INVITE. Are you using version 3.1.7 of the Sipura firmware? jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] SPA3000 as trunk - no caller ID
Kerry Garrison wrote: Upgraded to 3.1.7 Excerpts from Asterisk Log: Oct 27 07:43:50 DEBUG[1531]: cdr_mysql: SQL command as follows: INSERT INTO snip ... Here is a link to a screenshot of the SPA3000 settings: http://techdatapros.com/temp/spa3000.gif I get connection refused at that URL. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] SPA3000 as trunk - no caller ID
Kerry Garrison wrote: A phone plugged into it will grab the CID on about the second ring and I have adjusted the SPA3000 out to 5 rings with no difference. What gets passed to asterisk is whatever is set in the 3000's Display Name field. If the Display Name field is blank, then nothing comes across and the phones display 'Unknown'. I have been wondering if there is a variable you can put into the display field. There are some fields that use variables like $PROXY and $NOTIFY, of course simply trying $CID uses '$CID' as the caller ID. You don't need any clever manipulation tricks with the current firmware. Have you got PSTN CID for VOIP CID set to yes ? jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] SPA3000 as trunk - no caller ID
Kerry Garrison wrote: Yes I do have that set to Yes. Does the SPA-3000 show the caller ID in the last call field in the summary page ? It's capable of interpreting a bewildering array of callerid schemes - is it set to what your local telco is generating ? jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] E1/T1 failover hardware
Warning ! I know zip about electronics. I've been looking for a device to handle the switching of an E1 connection from one Asterisk box to another in the event of a catastrophic server failure. All of the solutions I've seen so far have been designed to handle the situation where the telco line faults so that the local PBX can switch to a secondary E1. I've come across this application note : http://www.maxim-ic.com/appnotes.cfm/appnote_number/2857 which describes T1/E1/J1, N+1 Redundancy With Analog Switches These parts are obviously designed to be built into E1 boards - hence, I think, the protection circuitry. Here's the question, then : what (apart from jumping through regulatory hoops) is to stop a simple array of MOSFETS (and a bit of control circuitry) implementing a failover switch controlled (say) by a pin on a serial or parallel port ? jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] E1/T1 failover hardware
Sergio Serrano wrote: http://www.junghanns.net/en/ISDNguard_produkt.html srsergio -Mensaje original- De: John Daragon [mailto:[EMAIL PROTECTED] Enviado el: jueves, 20 de octubre de 2005 17:24 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] E1/T1 failover hardware Warning ! I know zip about electronics. I've been looking for a device to handle the switching of an E1 connection from one Asterisk box to another in the event of a catastrophic server failure. All of the solutions I've seen so far have been designed to handle the situation where the telco line faults so that the local PBX can switch to a secondary E1. Thanks Sergio. I won't need to get the soldering iron out after all. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] E1/T1 failover hardware
Jon Pounder wrote: Warning ! I know zip about electronics. why not just use a multipole relay ? a 4pole double throw relay gives you 4 sets of contacts for the 2x tx and 2x rx wires. if you want to control with a bit in a parallel port, use something like a uln2003 relay driver (if the coil current is low enough), or a couple discrete transistors with the right gain and power handling. use the 12vdc out of a spare drive connector to power the relay. I would use one relay rather than 2 dpdt ones so that the switches are mechanically locked together and if one relay sticks you don't get a weird combination of circuits connected. Nothing will break, and the phone cops won't likely bother you if this does happen, but it could be real annoying and hard to diagnose if it does. This is basically the electromechanical equivalent of you pulling one cable and plugging in another (which is what I was going to do with some T1 routers), except, I found the TXPort. Good idea. I just have an irrational dislike of moving parts. And I *like* MOSFETS ! This actually is meant for failing between telco circuits, but works just fine working failing between CPE instead. it actually has csus, reframers, clock generator etc, as well as the relay circuit I describe to do the switchover. it actually samples the lines and uses some intelligence to see which to switch to. The device is obsolete so you'll only find it surplus now, and its t1 only as far as I know but there is probably E1 gear around that does the same thing. I bought mine for $20 so it was not even worth thinking about my own setup for that price, but they were listed at up to $3000 when new. I've only had a quick look for these, but E1 ones seem to be thin on the ground and expensive, and I have a horrible feeling that all the reframing stuff just adds another set of variables if something goes wrong somewhere. Thanks again. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] chan_capi-0.6 configuration Query with Eicon Diva 4BRI
Voicomm User wrote: Hello Hardware: Eicon Diva 4BRI ISDN Card Software : Asterisk : Asterisk CVS-v1-0-08/13/05-19:51:52 Chan Capi: chan_capi-0.6 We are using an Eicon 4BRI ISDN Card here in Australia with Asterisk, connected to 4 OnRamp services with Telstra. There are 8 available channels, but after upgrading to latest capi driver we notice that the box is not able to handle more than 2 calls at the same time. An engaged signal is heard at the other end. After this happens once, some calls fail even when all channels are free. I don't see any messages on console for failes calls. Even when I turn on 'capi debug' and 'set verbose 20'. The telstra personnel have confirmed busy signal is sent out by the PABX. But its bizarre not to see any messages. No error messages are logged as well. capi info : Contr1: 2 B channels total, 2 B channels free. Contr2: 2 B channels total, 2 B channels free. Contr3: 2 B channels total, 2 B channels free. Contr4: 2 B channels total, 2 B channels free. capi.conf [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [isdn] isdnmode=ptp ; Is this correct for Point to Point Mode? msn=8 digit local number group=1 incomingmsn=* controller=1,2,3,4 ; there are 4 controllers devices=2 ; should this be 8? softdtmf=on relaxdtmf=on accountcode= context=main-menu echocancelold=yes ;echocancel=yes ; Turning this on gives a error message each time a call is terminated. usecallerid=yes callerid=asreceived ;echosquelch=1 ;echotail=64 ;callgroup=1 ;pickupgroup=1 The syntax has changed a bit. Time was when the devices= line basically said OK, that's this controller done with, let's commit that and start on the next one... With 0.6 (if I read it correctly) it goes : [general] . . [some_string] group=1 isdnmode=did -- note this has changed [DID/MSN] incomingmsn=* rxgain=1.0 txgain=0.8 controller=1 softdtmf=0 accountcode= context=from-pstn echosquelch=0 echocancel=yes echotail=64 devices=2 [some_other_string] group=1 isdnmode=did incomingmsn=* rxgain=1.0 txgain=0.8 controller=2 softdtmf=0 accountcode= context=from-pstn echosquelch=0 echocancel=yes echotail=64 devices=2 Hope this helps... jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] chan_capi configuration with AVM C2 card
Hi; I've been asked to take a (remote) look at an [EMAIL PROTECTED] system running asterisk 1.0.9 on Centos 3.5. It's running chan_capi-0.3.5 It has an AVM c2 ISDN card which is plugged into what I believe to be a couple of BT ISDN2e System Access (i.e. point to point) connections. We've placed a support call to BT to find out how these lines are actually provisioned, but had no response so far. I'd be grateful if anyone could shed some light on what we're seeing : We've loaded the C2 firmware, and the /etc/capi.conf reads C2 c2.bin DSS1 - - - - P2P C2 - DSS1 - - - - P2P We see no traffic if P2P is not set in /etc/capi.conf. Surprisingly (to me, at least) if I set isdnmode=ptp in /etc/asterisk/capi.conf asterisk will not pick up incoming calls. Incoming ISDN calls *are* answered when isdnmode=ptm is set. We have not yet been able to negotiate an outgoing call. Each attempt is rejected by the telco with a reason code of 0x3302. Here's the dial command : exten = s,1,Dial(CAPI/xx:b$OUTNUM$) and the capi.conf : [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=xx isdnmode=ptm incomingmsn=* controller=1 softdtmf=1 accountcode= context=from-pstn echosquelch=0 echocancel=yes echotail=64 devices=2 msn=xx isdnmode=ptm incomingmsn=* controller=2 softdtmf=1 accountcode= context=from-pstn echosquelch=0 echocancel=yes echotail=64 devices=2 Here's the CAPI trace of the outgoing call : Oct 6 20:04:17 VERBOSE[12454]: == found capi with omsn = xx Oct 6 20:04:17 VERBOSE[12454]: == CAPI Call CAPI[contr2/xx]/1 with B3Oct 6 20:04:17 VERBOSE[12454]: == CAPI Call CAPI[contr2/xx]/1 with B3-- c reating pipe for PLCI=-1 Oct 6 20:04:17 VERBOSE[12454]: sent CONNECT_REQ MN =0x6 Oct 6 20:04:17 VERBOSE[12454]: -- Called xx:byy Oct 6 20:04:17 VERBOSE[12454]: -- CONNECT_CONF ID=001 #0x0006 LEN=0014 Controller/PLCI/NCCI= 0x202 Info= 0x0 Oct 6 20:04:17 VERBOSE[12454]: -- CONNECT_CONF ID=001 #0x0006 LEN=0014 Controller/PLCI/NCCI= 0x202 Info= 0x0 Oct 6 20:04:17 VERBOSE[12454]: == received CONNECT_CONF PLCI = 0x202 INFO = 0 Oct 6 20:04:23 VERBOSE[12454]: -- DISCONNECT_IND ID=001 #0x0003 LEN=0014 Controller/PLCI/NCCI= 0x202 Reason = 0x3302 Oct 6 20:04:23 VERBOSE[12454]: == DISCONNECT_IND PLCI=0x202 REASON=0x3302 Oct 6 20:04:23 VERBOSE[12454]: sent DISCONNECT_RESP PLCI=0x202 Oct 6 20:04:23 VERBOSE[12454]: -- CAPI Hangingup TIA jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] [Sorta OT] Eicon DIVA with [EMAIL PROTECTED]
Hi;
I've got an AAH installation where a customer wants to install an active
Eicon DIVA BRI card. AAH is built on Centos 3.5 which is currently at
kernel 2.4.21.37. Support for Eicon active cards is built-in.
I've debugged and run the [EMAIL PROTECTED] install-Eicondiva script but when I try to
run divactrl load -c 1 -f ETSI -Debug I get a response :
A: can't get card type for DIVA adapter number 1
dmesg reveals that lincfg.c has reported the error :
DIVA Server Driver - initialising
DIVA Server Driver - Version 2.0.16
Divas: DIVA Server BRI (U) Found
Divas: DIVA I/O Base already in use 0xf100-0xf11f
Divas: 0 cards detected
Divas: Not loaded
(here's the code that produces it ...)
if (check_region(Card.io_base, 0x20))
{
printk(KERN_WARNING Divas: DIVA I/O Base already in use
0x%x-0x%x\n, Card.io_base, Card.io_base + 0x1F);
wDeviceIndex++;
continue;
}
cat /proc/iomem shows :
f020-f020 : PCI device 1133:e013 (Eicon Technology Corporation)
f021-f02100ff : PCI device 1133:e013 (Eicon Technology Corporation)
f100-f1ff : PCI device 1133:e013 (Eicon Technology Corporation)
Has anyone else out there got an Eicon DIVA card running under AAH ? Or
have any idea why this is happening ?
TIA.
jd
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Re: [Asterisk-Users] [Sorta OT] Eicon DIVA with [EMAIL PROTECTED]
Armin Schindler wrote:
On Sun, 2 Oct 2005, John Daragon wrote:
Hi;
I've got an AAH installation where a customer wants to install an active Eicon
DIVA BRI card. AAH is built on Centos 3.5 which is currently at kernel
2.4.21.37. Support for Eicon active cards is built-in.
I've debugged and run the [EMAIL PROTECTED] install-Eicondiva script but when I
try to run
divactrl load -c 1 -f ETSI -Debug I get a response :
A: can't get card type for DIVA adapter number 1
dmesg reveals that lincfg.c has reported the error :
DIVA Server Driver - initialising
DIVA Server Driver - Version 2.0.16
Divas: DIVA Server BRI (U) Found
Divas: DIVA I/O Base already in use 0xf100-0xf11f
Divas: 0 cards detected
Divas: Not loaded
(here's the code that produces it ...)
if (check_region(Card.io_base, 0x20))
{
printk(KERN_WARNING Divas: DIVA I/O Base already in use 0x%x-0x%x\n,
Card.io_base, Card.io_base + 0x1F);
wDeviceIndex++;
continue;
}
It seems AAH is using the old driver in kernel. This driver is not
maintained any more and produce errors. Use the current drivers from
www.melware.net or the source level RPM from Eicon.
Thanks Armin;
I'll do that...
jd
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Re: [Asterisk-Users] Integrating with existing analog PBX
Martin Allen wrote: Asking about inserting Asterisk between a 4 line analogue PBX and the outside world... The proposed solution (with 4 x FXO and 4 x FXS using 2 TDM400 cards) will work fine until the asterisk box dies or suffers power failure. An alternative may be to use 4 Sipura SPA-3000 ATAs (which have an FXO and an FXS port as well as an RJ45 network port (think of them as two ATAs an a single box...) and are cheap (see http://www.voiptalk.org ) PSTN ***|| * * SIP to FXO +---+ * Asterisk* -| | * * |SPA-3000 | * * -| | * * SIP to FXS +---+ ***|| PABX In the event of power failure the FXO port is switched directly to the FXS port, effectively bypassing the IP side of things completely. Actually, I think you *could* build what you're describing just with the SPA-3000s, but you would, of course, lose a lot of flexibility... jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Not enough lines available for Asterisk implemetation
Wayne Gemmell wrote: Hi all I am looking at implementing asterisk at a company with two ISDN bricks (60 lines). I know that the VoIP will absorb at least on brick worth of lines but that still leaves me with a need for 30 ISDN lines. As far as I can tell most of the Digicom cards have 4 FXS ports and I've read on this list that at most two could coincide in a box simultaneously without causing an interupt flood. 1) is my info okay so far? 2)What would be the best way for be to implement the other 22 lines? Is there hardware I'm not aware of? Wayne, hi; I guess what you're describing is 2 x ISDN30 connections (around 2Mbit/s each ?) I'm not familiar with the SA telephone system, but in the rest of the world (more or less) the card you'd need is the Digium TE110P which is switchable between T1 (24 channel) and E1 (30 channel) ISDN. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Which Linux distribution?
YT Lim wrote: We have tried Asterisk 1.0.9 on FC4 and have never been able to get CAPI (with Fritz card, fcpci) to work properly. Apart from that Asterisk works fine in switching internal calls. But it's useless if we can't make outgoing calls on our ISDN line. We are considering abandoning FC4 for Debian or SuSe. What is the general concensus on the best Linux to run Asterisk with CAPI? SUSE (as far as I know) is the only distro that really *expects* you to be using ISDN2e as a matter of course. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Which Linux distribution?
Tzafrir Cohen wrote: On Wed, Sep 07, 2005 at 10:10:05AM +0100, John Daragon wrote: YT Lim wrote: We have tried Asterisk 1.0.9 on FC4 and have never been able to get CAPI (with Fritz card, fcpci) to work properly. Apart from that Asterisk works fine in switching internal calls. But it's useless if we can't make outgoing calls on our ISDN line. We are considering abandoning FC4 for Debian or SuSe. What is the general concensus on the best Linux to run Asterisk with CAPI? SUSE (as far as I know) is the only distro that really *expects* you to be using ISDN2e as a matter of course. Only Linux distro that is generally something that is a bit hasty to say, given the fact that there are so many of them ;-) . You're absolutely right. Mandrake is quite Europe-centric as well. I'm not sure about ISDN support. It's shipped with the packages; I looked at it when I first started installing *, but couldn't get fcpci to work at the time. CAPI appears to have been written on (or for) SUSE in the first place, and SUSE was the first distro I came across that supported ISDN2e out of the box. Debian has generally a large european installed base and a variety of ISDN-related packages as a result. Sorry, I won't make your life easier :-p You mean it's *supposed* to be easy ? jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Small office setup/using analog lines w/ Asterisk
jennyw wrote: Hi, We recently tried installing Asterisk for a small office. We figured the safest way to go would be to buy from someone who sold equipment specifically for Asterisk and to use a consultant that they recommended. However ... it didn't turn out so great. Sound quality is terrible -- the echo is pretty bad, and there are popping noises, too. Callers say that people on the Asterisk end sound very faint, while people on the Asterisk end hear people maybe too loundly (might be related to the popping noises -- sounds like when you have stereo turned up too high). The reseller and the consultant both say that the most likely cause for this is using Digium cards w/ analog phone lines. Apparently, they say, sound quality can be pretty bad. I called Digium and they gave me some suggestions for settings, but nothing has worked well. So I wanted to ask others ... has anyone had good luck with using analog phone lines and Asterisk? Especially with Digium cards (we use the TDM400P)? Although from reading articles on the net it sounds like people do have a lot of echo problems, it also sounds like some people are using analog phone lines with some success. FYI, what I've mainly done is try changing echotraining, echocancel, echocancelwhenbridged, txgain, and rxgain in zapata.conf. I've heard from the reseller that what might work better is to trade the Digium cards in for VegaStream gateway. It's more expensive, but apparently has a DSP built in that should increase voice quality. Of course, they say there are no guarantees with this. They also mentioned (after the fact) that Asterisk systems don't necessarily save money. So far, the experience has been very frustrating and I'd love to hear some success stories from others (or more info on what I can realistically expect from an Asterisk system)! And, of course, some ideas on how I can get things to work better. One of the next tests will be using Asterisk with a VoIP provider to see what the sound quality is like with digital on both ends. PRI sounds like it'd be even better, but for an office w/ 5 people, it sounds pretty expensive. How do other people do this? I started using Asterisk for my own small business about a year ago. Externally we have a single analogue PSTN line (it's the house one...), an ISDN2e connection and an IAX2 connection (over 20:1 256/512kbps ADSL) with a DID in central London. The analogue line comes in to an old X100P, and the ISDN into an AVM Fritz! passive card. Internally, we have a TDM400 which talks to analogue phones in the house. In my office (which is in a different building) we have a mixture of Snom and ipDialog phones and a Grandstream ATA attached to a fax machine. We get a little echo on the ipDialog phone (but not enough to be a problem) when we talk to people on analogue phones. One of the handsets attached to the TDM400 is a DECT phone, and there's a little flurry of training noise at the beginning of an incoming call, but after that the quality is good to perfect. I'm just beginning to sell Asterisk systems. I agree that for some installations, it doesn't really make economic sense. In the UK, at least, you have to fall into a specific band of numbers-of-users and minutes-per-month for IP telephony to show a saving. Some of the small 3-line-8-extension systems from (say) Panasonic will be cheaper than Asterisk once the hardware is bought and the time (or consultancy) applied. Of course, these systems don't have much in the way of flexibility or features, and I'm talking at the moment to a company that has three sites, is using Cisco's Call Manager, and has an Asterisk system merely to convert the H.323 from the Cisco to IAX2. In this case, * could replace the CCM system in its entirety. By the time you have 100 users, * is a no-brainer in economic terms. Small users only really save (IMHO) if they a) use an awful lot of minutes (or call abroad a lot), b) need flexibility of features, or c) need internal control. Of course there may be local or exceptional circumstances which make this all a load of rubbish ! YMMV. Oh, and on echo; read : http://lists.digium.com/pipermail/asterisk-users/2005-March/096754.html jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ 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] call load balancing
Jean-Michel Hiver wrote: Dave Redmore wrote: Hello All, Wondering what sort of real world mileage people are getting out of different internet connecions - i.e. different DSL connection speeds, cable modems, etc... Is it reasonable to hope to carry 10 - 15 concurrent calls on a 768K DSL? I'm not talking about theoretical BW or looking for any difinitive absolute guarantee... With DSL and Cable - there is no guarantee, so I'm wondering what folks are getting with real world usage... I do up to 10 g.729 channels over 1024 / 256 DSL without noticeable difference in call statistics (i.e. avg length of calls). If you are using ADSL, the maximum bandwith you'll be able to use is your upload rate since VoIP calls send data bidirectionally. Snip ... Jean-Michel, hi; Is that using SIP or IAX2 ? I'd assumed you'd be able to get more than that throughput out of an IAX2 trunk because of the sharing of RTP overhead ? jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ 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] Polycom distributor in the UK ?
Chris Mason (Lists) wrote: John Daragon wrote: Hi; I'm looking for a Polycom distributor in the UK who can supply a small number (around 20) IP301 / IP501 handsets. Can anyone recommend someone ? jd I have been buying from Zycko - very efficient and on the ball. Ta. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ 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] 12 seat call centre with Asterisk, VoIP only, UK - possible?
Mike Dent wrote: Hi, I've had an inquiry for a small UK call centre, mostly outbound calls. I get the impression they are mainly calling 3G mobile phones, monthly phone bill, with calls is approx £5,000 for several feature lines. How feasible is something like this with asterisk? I guess one big question is which type of circuit to use, ADSL in the UK is only 256kbs upstream, some providers do bonding but I'm not sure its supported fully by BT :( The other option is SDSL which is not too cheap! If most calls are to mobiles on *known* networks (and I know it may be difficult to work out which network a number is on, so this may not work for you ...) might it not be cheaper to get hold of a 3G gateway and route at least some of your calls directly over the relevant mblie network ? jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ 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] 2 minutes pause before ring on H323 channel
Peter Valkov wrote: John Daragon wrote: Peter Valkov wrote: I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as described in README file. Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11 I tested it with following phones: -- XLite (SIP softphone) -- QMix SIP IP phone (PA168F) -- SJPhone (H323 softphone) -- QMix H323 IP phone (PA168F) -- FireFly (IAX2 softphone) Everything works fine except a problem with h323 extension dialing. Behavior is the same for both SJPhone (soft phone) and QMix (PA168F). When I dial such extension I have to wait 2 minutes exactly (120 seconds) before extension rings. After long way of trial and errors with .conf files I managed to minimize this time to 1 minute exactly (60 seconds) exten = 20,1,Dial(H323/h323phone) ; this leads to 120 seconds pause before ring exten = 21,1,Dial(H323/h323phone at 192.168.0.101) ; this leads to 60 seconds pause before ring Peter, hi; I haven't looked at the openh323 code, and I might not get time to... but in my limited experience, 60 second delays are almost always DNS timeouts. Yep - down in openh323/src/transports.cxx there's a method H323TransportAddress::GetIpAndPorts() which is called (eventually) by MakeCallLocked(). This in turn calls GetPortByService() and GetHostByAddress(). My guess is that the 60 second wait is caused by a request to a DNS server that is never honoured. Of course, I've been wrong before... It is definitely DNS problem. The strange thing is that from command line everything works just fine. I can perform DNS and reverse DNS lookup without problem. Here follows my brutal workaround. In file pwlib/include/ptbuildopts.h is defined P_DNS 1 I changed it to P_DNS 0 ... after that recompiled pwlib openh323 and chan_h323 ... make install from asterisk home dir ... and voila ... no more 60 or (120) seconds delays. I suppose that this approach is quite graceless... because in this way entire openh323 DNS resolver is disabled... but this is the only way I managed to get it working I'm still looking for proper solution of the problem... so any help or advice will be appreciated Wow, that *is* brutal. Still, at least you're working for the moment... And another data point for the 60 seconds is *always* DNS rule ! I don't have h.323 installed here, so I'm of limited utility for testing. What I would do, I think, is to perform an ethereal trace on requests to port 53. This is simple and will tell you whether the problem is inside or outside the asterisk machine. As DNS appears to be working otherwise, I'll have a look at the h.323 code again if I get the time today, just in case there's something obvious going on. Oh, and would it be possible for you to post your resolv.conf ? jd ___ 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] 2 minutes pause before ring on H323 channel
Peter Valkov wrote: I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as described in README file. Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11 I tested it with following phones: -- XLite (SIP softphone) -- QMix SIP IP phone (PA168F) -- SJPhone (H323 softphone) -- QMix H323 IP phone (PA168F) -- FireFly (IAX2 softphone) Everything works fine except a problem with h323 extension dialing. Behavior is the same for both SJPhone (soft phone) and QMix (PA168F). When I dial such extension I have to wait 2 minutes exactly (120 seconds) before extension rings. After long way of trial and errors with .conf files I managed to minimize this time to 1 minute exactly (60 seconds) exten = 20,1,Dial(H323/h323phone) ; this leads to 120 seconds pause before ring exten = 21,1,Dial(H323/[EMAIL PROTECTED]) ; this leads to 60 seconds pause before ring Peter, hi; I haven't looked at the openh323 code, and I might not get time to... but in my limited experience, 60 second delays are almost always DNS timeouts. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ 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] 2 minutes pause before ring on H323 channel
John Daragon wrote: Peter Valkov wrote: I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as described in README file. Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11 I tested it with following phones: -- XLite (SIP softphone) -- QMix SIP IP phone (PA168F) -- SJPhone (H323 softphone) -- QMix H323 IP phone (PA168F) -- FireFly (IAX2 softphone) Everything works fine except a problem with h323 extension dialing. Behavior is the same for both SJPhone (soft phone) and QMix (PA168F). When I dial such extension I have to wait 2 minutes exactly (120 seconds) before extension rings. After long way of trial and errors with .conf files I managed to minimize this time to 1 minute exactly (60 seconds) exten = 20,1,Dial(H323/h323phone) ; this leads to 120 seconds pause before ring exten = 21,1,Dial(H323/[EMAIL PROTECTED]) ; this leads to 60 seconds pause before ring Peter, hi; I haven't looked at the openh323 code, and I might not get time to... but in my limited experience, 60 second delays are almost always DNS timeouts. Yep - down in openh323/src/transports.cxx there's a method H323TransportAddress::GetIpAndPorts() which is called (eventually) by MakeCallLocked(). This in turn calls GetPortByService() and GetHostByAddress(). My guess is that the 60 second wait is caused by a request to a DNS server that is never honoured. Of course, I've been wrong before... jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ 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] UK ISDN with Asterisk
Henry Owens wrote: Hi, Get yourself a 'Fritz!Card PCI' - also marketed by BT themselves as a 'BT Speedway ISDN' adapter - these seem to be the most cheap and supported of low-end ISDN2 adapters Will do - they seem pretty inexpensive (even for the BT Speedway card is only about £35). From doing a bit of poking, SuSE 9.1 seems to be the latest OS for which drivers are available. Is anyone using one of these cards successfully, and if so, on SuSE? One more question (and probably a pretty basic one, but i'm not that familiar with PSTNs) - will i need two of these cards in order to use both channels? Looking forward to getting this going now, and much more confident, thanks for your support! Henry, hi; I'm running a small Asterisk PABX under SuSE 9.1. I have one analogue (PSTN) line, a single ISDN2e connection (i.e. 2 channels - one adaptor card) and a London PSTN number which gets routed to me via IAX, and I support 2 internal SIP phones and 4 internal analogue handsets. DID and whatever CLID is called in ISDN work fine. CLI on the analogue line is a nightmare because the Digium hardware doesn't support BT's CLI, so I have a modem picking that up and inserting it into Asterisk with (so far) variable results. Outgoing calls go either via the landlines, or via the Docklands-terminated IAX channel. All works pretty well - looks like just the sort of solution your client may need. Do be aware that supporting multiple ISDN2e cards might problematic. Not impossible, but problematic... jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ 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] UK ISDN with Asterisk
Henry Owens wrote: John, Thanks very much for the detailed response, that sounds pretty much like what i'm looking for (1x BT ISDN2e and 1x analogue). Are you using one of the Digium 4 port BRI cards, or what hardware are you using? I'm using an AVM Fritz card with chan_capi. They're pretty cheap on eBay if you're suffering sticker shock... Of course, they're not as efficient as the active ones, but they're a lot cheaper and you already own the PC, I guess. It would be my intention to use the ISDN primarily for incoming, and VoIP for outgoing to cut costs, and increase functionality. You mentioned your PSTN number is routed to you via IAX; can that number be included in local directories? H... Probably not. I'm using vioptalk.org's Prepay Silver which allocates a geographic number. But it's *voiptalk's* number as far as the network is concerned, so I've no idea how you'd get it into a directory. I chose this because, although the company is ex-directory, I want people to be able to phone back so I'll show the geographic number in outgoing CLI. Some providers allow you to specify your own CLI on outgoing SIP, if that's any use. I don't think the analogue CLI should be a problem, since the ISDN should be taking most of the incoming calls. Does CLI work ok on the ISDN? Oh, yes. Of course, it's not *quite* the same. Here's an example : Analogue CLI :01460 234068 ISDN CLI*:441460234068 So if you want to call people back, you're going to have to play with extensions.conf... * Yes, I *know* that's not what it's really called... At this point i would intend to use only 1 ISDN card, so i'll cross the multiple card bridge when (and if) i come to it. OK. Drop me an email if you need any help (bearing in mind that paying clients get first dibs on my time!). jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ 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
[Asterisk-Users] Fedora Core 2 (kernel 2.6.5), CAPI and Fritz PCI
I have an Asterisk box running happily under Fedora Core 2 with a X100P and a TDM400P, and now I'd like to integrate it to my ISDN2e connection using either an AVM Fritz PCI card or an Eicon DIVA passive card, both of which I have sculling around. I've successfully used the AVM card under RedHat 8.0, but I'm having difficulty finding information on running it under the Fedora 2.6 kernel. Is there anyone out there running this combination ? jd -- John Daragon argv[0] limited [EMAIL PROTECTED] Lambs Lawn Cottage, Staple Fitzpaine, Taunton TA3 5SL, UK (house) 01460 234537 (office) 01460 234068 (mobile) 07836 576127 (fax) 01460 234069 ___ 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
