Re: [asterisk-users] While the VoIP-Info.org site is down...
[Al Bochter wrote on 15/03/2007 12:25 PM]: So does anyone know when Voip-info.org will be back up? There is a message on the list from James Thompson with the subject voip-info.org status update saying it suffered a major hard drive crash and should be back tomorrow. Looking at the headers that message was sent two hours before yours. Regards Darryl ___ --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] FW: failure notice
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Paul wrote: Anyway, bumping him is not extreme at all. IIRC - some lists are setup to automatically unsubscribe people after N days of delivery failures. We only see this individually when we post but the list server is probably getting this for every new post to the list. Very unlikely. If the person sending the post is getting the bounce message, then the list server is *not*. Therefore it won't know that the subscriber is bouncing. Regards Darryl -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.1 (MingW32) iD8DBQFCidoR/XQ6DbmPjokRAhdxAJ49mb1Xvk/w78Hk4bMJwZF8ScbpgwCdFQLc KYAUNIJh620gDU+u+SghkPo= =KvXu -END PGP SIGNATURE- ___ 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] Cisco device for gatewaying SIP to H323 suitable for ~50 simultanious calls
Hey All, Our upstream provider requires the use of H323 and after several months (6!) of having problems with OH323 I've decided it might be worth biting the bullet and getting a cisco device that can gateway up to approximately 50 calls from SIP to H323. Would a 2500 or 2600 series do the job? Once we get to the point of 50 simultaneous calls hopefully we'll be able to get something bigger. Needs to support incoming and outgoing calls. TIA Darryl ___ 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] Cisco device for gatewaying SIP to H323 suitable for ~50 simultanious calls
Hey All, Our upstream provider requires the use of H323 and after several months (6!) of having problems with OH323 I've decided it might be worth biting the bullet and getting a cisco device that can gateway up to approximately 50 calls from SIP to H323. Would a 2500 or 2600 series do the job? Once we get to the point of 50 simultaneous calls hopefully we'll be able to get something bigger. Needs to support incoming and outgoing calls. TIA Darryl ___ 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] No outgoping calls with ISDN
Paulo Adriano wrote:
I need some help from you. I´m using Isdn4linux with Asterisk and
incoming calls are working but anytime I whant to make an outgoing call
I also use isdn4linux for interfacing with my BRI line. I have a macro
set up for the actual dialling:
-- start
[macro-isdnout]
; ${ARG1} - Device to ring out on
; ${ARG2} - Number to dial
exten = s,1,Dial(${ARG1}:${ARG2},,r)
-- end --
And then I call it like this:
-- start
exten = _[08]XXX.,1,Macro(isdnout,Modem/g0,${EXTEN})
exten = _1*XXX.,1,Macro(isdnout,Modem/g1,${EXTEN:2})
-- end --
That means that if I dial a number starting with a 0 or an 8 then it
uses the g0 group in the modems.conf and if I dial a number starting
wiht a 1* then it uses the g1 group I set up in modems.conf. I set 2
groups up so that I could choose which MSN to use for the outbound
callerid (one for my 'Home' number and one for my 'Office' number).
HTH.
Regards
Darryl
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Re: [Asterisk-Users] BRIDDI
On Thu, 2 Sep 2004 14:51:10 +0200, Altus Snyman [EMAIL PROTECTED] wrote: Good day all Is there anyone who has experience with ISDN BRIDDI? I'm currently using a BRI with ISDN4Linux. I want to know if asterisk can distinguish between the different numbers? Yes, it can differentiate between the numbers assigned to my service. I want each number to play a different intro/answering message? As long as you set up your dial plan correctly in asterisk, you can do this. Regards Darryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 Codec on Max OS X
Hey All, Just wondering if there is a version of the G729 Codec available for Mac OSX? I can see almost all the x86 infrastructures ... Regards Darryl -- Darryl Ross Senior Network Engineer OEG Australia Email: [EMAIL PROTECTED] Phone: 08 81228363 Office: 08 81226361 If you want to live up to the whole There is more than one way to do it slogan, you have to give someone a swiss army chainsaw ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] desparate for help DEV LITE KIT
steve wrote: I'm baffled. All I want is a simple 1x1 PBX to keep telemarketers from ringing my phone. If I can't get this working I'm having my phone disconected. lol Surely the shrill tone would be good for keeping telemarketers away? ;) -- Darryl Ross Senior Network Engineer OEG Australia Email: [EMAIL PROTECTED] Phone: 08 81228363 Office: 08 81226361 If you want to live up to the whole There is more than one way to do it slogan, you have to give someone a swiss army chainsaw ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zap show channels - no such command
Hi Imran, I seem to have done the zaptel installation - what am I missing - i still don't have a chan_zap.so file? Did you rebuild Asterisk after installing Zaptel? chan_zap.so is built as part of the actually Asterisk build, AFAIK. added chan_zap.so to modules.conf, when asterisk starts up it can't find it. If you don't have Zaptel installed when you build Asterisk, it might not build chan_zap.so. On my system the asterisk modules are in /usr/lib/asterisk/modules/. Regards Darryl -- Darryl Ross Senior Network Engineer OEG Australia Email: [EMAIL PROTECTED] Phone: 08 81228363 Office: 08 81226361 If you want to live up to the whole There is more than one way to do it slogan, you have to give someone a swiss army chainsaw ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Festival Issues
Hey All, I now have Festival compiled, installed and running using the instructions on the Wiki page. When I try to change the voice that is being used however, I am running into a problem. I get the following in the festival server log: Cannot open file /tmp/est_10877_0/utt.wav as tokenstream Wave load: can't open file /tmp/est_10877_0/utt.wav Cannot load wavefile: /tmp/est_10877_0/utt.wav When I look in the /tmp/est_10877_0 folder, while the sound file is still playing according to Asterisk, the following seems to be created: total 56 drwxr-xr-x2 darryl users4096 Aug 19 16:28 . drwxr-xr-x 29 root root 4096 Aug 19 16:28 .. -rw-r--r--1 darryl users 0 Aug 19 16:28 tmp.f0 -rw-r--r--1 darryl users 22365 Aug 19 16:28 tmp.lab -rw-r--r--1 darryl users 0 Aug 19 16:28 tmp.mcep -rw-r--r--1 darryl users 0 Aug 19 16:28 tmp.raw -rw-r--r--1 darryl users 23411 Aug 19 16:28 utt.feats I've tried setting a number of different voices. Some don't give any audio and some only seem to give a small 'blip' of sound. Does anyone have any ideas? Thanks in advance Darryl -- Darryl Ross Senior Network Engineer OEG Australia Email: [EMAIL PROTECTED] Phone: 08 81228363 Office: 08 81226361 If you want to live up to the whole There is more than one way to do it slogan, you have to give someone a swiss army chainsaw ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerId
Simon Brown wrote: In my zapata.conf, I have callerid=unknown That doesn't look right to me. Try: callerid=Unknown Cheers Darryl -- Darryl Ross Senior Network Engineer OEG Australia Email: [EMAIL PROTECTED] Phone: 08 81228363 Office: 08 81226361 If you want to live up to the whole There is more than one way to do it slogan, you have to give someone a swiss army chainsaw ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to purchase ISDN (BRI) cards in Australia (preferably)
Hey Shaun, Because of the Isdn4Linux DTMF issue, I don't want one of those cards. I've already spent too much time messing about with my current card. I have a Traverse NetJet card which I got working without any problems at all. I'm pretty sure all I had to do in order to get DTMF working all I had to do was set my SIP client to use rfc2833 signalling. I'm not at home at the moment, but I can check that when I get home tonight. The only thing I have not been able to get working is SpanDSP for receiving incoming faxes. On another note, if anyone has any NetJet cards they don't want, I'll be happy to take them :) Cheers Darryl -- Darryl Ross Senior Network Engineer OEG Australia Email: [EMAIL PROTECTED] Phone: 08 81228363 Office: 08 81226361 If you want to live up to the whole There is more than one way to do it slogan, you have to give someone a swiss army chainsaw ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Festival Installation - Asterisk 1.0-RC2 Debian Woody
Hey All, Thought I'd take a bash at trying to get Festival to work here on my lab system with the aim of using it to create our IVR menu prompts. I've spent most of the afternoon searching through the Wiki, the Festival website and Google and I've got a couple of questions. First one is that the 'Asterisk+festival+installation' page on the Wiki mentions the RedHat 9 RPMs for Festival 1.4.2 will not work with asterisk. Is this also the case with the deb files in Woody? (1.4.2-2.1) I have tried to use the debian provided package, but I get the following in the Asterisk console (I've obtusified the phone IP address...): -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hd' -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created in state: Down -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '*' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '1' -- Endpoint 'aaln/[EMAIL PROTECTED]' observed '4' -- Executing Answer(MGCP/aaln/[EMAIL PROTECTED], ) in new stack -- MGCP mgcp_answer(MGCP/aaln/[EMAIL PROTECTED]) on aaln/[EMAIL PROTECTED] -- Executing Festival(MGCP/aaln/[EMAIL PROTECTED], This is only a test) in new stack == Parsing '/etc/asterisk/festival.conf': Found Aug 18 16:28:17 WARNING[491537]: app_festival.c:440 festival_exec: Festival returned ER == Spawn extension (local-clients, *14, 2) exited non-zero on 'MGCP/aaln/[EMAIL PROTECTED]' -- Executing Hangup(MGCP/aaln/[EMAIL PROTECTED], ) in new stack == Spawn extension (local-clients, h, 1) exited non-zero on 'MGCP/aaln/[EMAIL PROTECTED]' -- No command found on [203.33.246.xx] for transaction 168. Ignoring... -- Endpoint 'aaln/[EMAIL PROTECTED]' observed 'hu' -- MGCP handle_request(aaln/[EMAIL PROTECTED]) ast_channel already destroyed -- MGCP handle_request(aaln/[EMAIL PROTECTED]) set vmwi(-) In the festival server log I get: client(8) Wed Aug 18 16:28:17 2004 : accepted from localhost client(8) Wed Aug 18 16:28:17 2004 : disconnected Assuming that the debian packages are not compatible, which version of Festival do I need? The Wiki page mentioned above says to grab the tarball of 1.4.3, which is no longer available from the website. Only 1.95 is available. Will that work? Does it need the patch mentioned on the Wiki page? Does anyone have an apt compatible repository of everything needed to get Festival working with Asterisk under Debian Woody? Thanks in advance Darryl -- Darryl Ross Senior Network Engineer OEG Australia Email: [EMAIL PROTECTED] Phone: 08 81228363 Office: 08 81226361 If you want to live up to the whole There is more than one way to do it slogan, you have to give someone a swiss army chainsaw ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival Installation - Asterisk 1.0-RC2 Debian Woody
Sebastian Sporleder wrote: Darryl Ross wrote: Assuming that the debian packages are not compatible, which version of Festival do I need? The Wiki page mentioned above says to grab the tarball of 1.4.3, which is no longer available from the website. Only 1.95 is available. Will that work? Does it need the patch mentioned on the Wiki page? Of course it is available on theri website! Have a look here: http://festvox.org/packed/festival/ I have installed it from souce yesterday and there are also patches for 1.4.1 and 1.4.2 in the CVS contrib folder! Doh, I must have missed that one. The site I found for it was at http://www.cstr.ed.ac.uk/projects/festival/ Thanks! Darryl -- Darryl Ross Senior Network Engineer OEG Australia Email: [EMAIL PROTECTED] Phone: 08 81228363 Office: 08 81226361 If you want to live up to the whole There is more than one way to do it slogan, you have to give someone a swiss army chainsaw ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CID on internal extensions
Ronan wrote: That does not work for me. I had tried that, but no luck. This is what I have in there for it. context=darby usecallerid=yes musiconhold=default echotraining=yes echocancel=yes signalling=fxo_ks channel = 19 callerid=119 I'm pretty sure with zaptel channels you configure a template, then assign the template to the channel. Eg, the channel = line needs to be the last line for that configuration. Try putting the callerid = before the channel = Regards Darryl -- Darryl Ross Senior Network Engineer OEG Australia Email: [EMAIL PROTECTED] Phone: 08 81228363 Office: 08 81226361 If you want to live up to the whole There is more than one way to do it slogan, you have to give someone a swiss army chainsaw ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe
Hi Mitul, I have just started using * and have been trying to set up MeetMe. So far I have not been able to start a conference. When I dial the conference extension (I am using X-Lite softphone), the call hangs up. I am not using Zaptel cards so I uncommented the ztdummy in the Zaptel Makefile. My configuration is as follows: I've only just looked at getting MeetMe going this week myself, so it's fresh in my memory :) You said you uncommented the ztdummy modules in the zaptel Makefile. Did you then compile the zaptel stuff, install it and configure it as per the directions at http://www.voip-info.org/wiki-Asterisk+timer+ztdummy ? My config, which worked fine, looks like: -- /etc/asterisk/meetme.conf conf = 100 -- /etc/asterisk/extensions.conf exten = 315,1,Wait(1) exten = 315,2,MeetMe(100|Ms) The 'M' option provides music on hold to the first user who enters the conference and the 's' option gives the user the ability to enter a config menu, so they can mute and unmute the audio they are sending. One thing I haven't really been able to find out is how the /etc/zaptel.conf and /etc/asterisk/zapata.conf is meant to be configured for ztdummy. I just left them at the default files. Hope that helps. Regards Darryl -- Darryl Ross Senior Network Engineer OEG Australia Email: [EMAIL PROTECTED] Phone: 08 81228363 Office: 08 81226361 If you want to live up to the whole There is more than one way to do it slogan, you have to give someone a swiss army chainsaw ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-wide variables
Hi Rene, As an example: I set the gateway's telephone number both in extensions.conf, oh323.conf and phones.conf but I don't want to change the same data in three separate files whenever I want to set a different telephone number for the gateway. I have no idea whether this would work or not, but couldn't you do a #include vars.conf in the relevant places in extensions.conf, oh323.conf and phones.conf and then define the variables in vars.conf ?? -- Darryl Ross Senior Network Engineer OEG Australia Email: [EMAIL PROTECTED] Phone: 08 81228363 Office: 08 81226361 If you want to live up to the whole There is more than one way to do it slogan, you have to give someone a swiss army chainsaw ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
Hi Francis, so no need to make a special dialplan to acomodate the weird numbering system we have in Brazil (sometimes we dial 7 numbers, sometimes 8, sometimes 12, sometimes 13, etc.) Actually, we also have non-fixed phone numbers in Germany. I think this is not weird, I think this is very good. And again, Asterisk supports this. Oh, so I how does Asterisk knows when to start dialing out the numbers, if there are no rules? Have a look at http://www.voip-info.org/wiki-Asterisk+Extension+Matching Regards Darryl -- Darryl Ross Senior Network Engineer OEG Australia Email: [EMAIL PROTECTED] Phone: 08 81228363 Office: 08 81226361 If you want to live up to the whole There is more than one way to do it slogan, you have to give someone a swiss army chainsaw ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk CDR UniqueID
Thanks for the reply Freddi, The problem I am having is I'm trying to work out how to link the CDR records into a single 'call stream', rather than having separate records per machine the call passes through. I had the same problem. One solution is to include the ip-adr or uname of the gateway that serves the call, then you can have a true unique-id. I did patch cdrcsv.c to include an exstra field which is the uname of my * server. Another possible way today is to use the 'SetCDRUserField' to let the 'uname' into your cdrstream. I've done some testing in setting both the account code and the CDR user field, but neither of them are being transferred between the two machines over the IAX2 link. Is there any way of setting a variable or some such thing and having it transfered over the IAX2 link? I've thought about playing with the CallerID field which we're not really using at the moment, but we probably will in the future. Any ideas? Cheers Darryl -- If you want to live up to the whole There is more than one way to do it slogan, you have to give someone a swiss army chainsaw ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk CDR UniqueID
Hey All, We are running a small SIP/IAX termination service at the moment (planning on growing it) with 2 asterisk machines. One terminates the SIP/IAX calls from our customers and one is our gateway to our upstream provider. Both machines are logging CDR data to the same postgres table using the cdr_psql module. The problem I am having is I'm trying to work out how to link the CDR records into a single 'call stream', rather than having separate records per machine the call passes through. Reading around on the Wiki and doing a bit of googling, I've worked out that what I'm trying to do is the normalization step of CDR mediation, but I have not been able to find out any specifics about how to go about it. I would have thought that the originating asterisk machine would generate the UniqueID for the call (Message-ID in SMTP terms) and pass that along the call path, but each machine is using the epoch timestamp of when it sees (or records, not sure which) of the call. I know I can use the NoCDR app on the first machine in the chain, but that does not scale if we need to add more machines to our network. Does: a) anyone have any idea of what I'm trying to explain, and b) have any pointers of where I can find more information about doing this? Thanks Darryl -- If you want to live up to the whole There is more than one way to do it slogan, you have to give someone a swiss army chainsaw ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P only dials a single digit
Hey All, Having a bit of a problem with a Wildcard X100P card. When I try to make an outbound call using the card, it picks the line up and then only dials a single digit. I've confirmed it's only dialling a single digit by listening on a phone plugged into a parallel socket. Incoming calls work fine, it's only outbound calls I'm having a problem with. I've tried searching the Wiki and haven't had much luck with Google (could be my search terms) I am in Australia and as far as I know I've set up the indications and everything ok. My /etc/zaptel.conf looks like the following: --- fxsks=1 loadzone=au defaultzone=au --- My /etc/asterisk/zapata.conf file looks like this: --- [channels] context=default usecallerid=no echocancel=yes rxgain=6.0 txgain=5.3 group=0 signalling=fxs_ks context=incoming channel = 1 --- My /etc/asterisk/indications.conf looks like this: --- [general] country=au [au] description = Australia ringcadance = 400,200,400,2000 dial = 425*25 busy = 400/375,0/375 ring = 425*25/400,0/200,425*25/400,0/2000 congestion = 400/375,0/375 callwaiting = 425/100,0/100,525/100,0/4700 dialrecall = !425*25/100!0/100,!425*25/100,!0/100,!425*25/100,!0/100,425*25 record = 1400/425,0/14525 info = 400/2500,0/500 --- When I try to make an outbound call through the card, I get the following in the asterisk console: --- -- Accepting AUTHENTICATED call from 203.33.246.1, requested format = 256, actual format = 8 -- Executing Dial([EMAIL PROTECTED]:4569]/1, Zap/g0|82814621|rtT) in new stack -- Called g0 -- Zap/1-1 answered [EMAIL PROTECTED]:4569]/1 -- Hungup 'Zap/1-1' == Spawn extension (localpstn, 82814621, 1) exited non-zero on '[EMAIL PROTECTED]:4569]/1' -- Hungup '[EMAIL PROTECTED]:4569]/1' --- Does anyone have any ideas what might be causing this behaviour? Thanks in advance Darryl -- If you want to live up to the whole There is more than one way to do it slogan, you have to give someone a swiss army chainsaw ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bri solution for Asterisk
Mark Elkins wrote: going to i4l means... incoming sound sometimes gets interpreted as DTMF - and when your caller humms a '#' - transfer kicks in... Outgoing DTMF simply does not work. (Don't do i4l!) It doesn't? Funny, no one must have told my NetJet card that -- it works fine! Regards Darryl -- If you want to live up to the whole There is more than one way to do it slogan, you have to give someone a swiss army chainsaw ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RC1 Mirror, was Re: [Asterisk-Users] Asterisk-1.0 RC1
Hi List I have also saved a copy, available at http://mirror.afoyi.com/asterisk/, which should be very quick for anyone in Australia. Regards Darryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Book
Steven Critchfield wrote: On Thu, 2004-07-08 at 09:16, [EMAIL PROTECTED] wrote: I do not recall telling anyone 6 weeks, My book located at www.saww.net/asterisk/ is being shipped to everyone that has not received their orders as of next week. maybe next time you should get your facts straight before lieing in this mailing list I do not have a problem that you are trying to write your own book all the best wishes but lies do not help Be aware that the URL you just posted says it is Backordered, ships in 1-3 weeks. That page has been updated since I first looked at it (it now has the table of contents listed). When I first looked at it, the availablity note __DID__ say Backorded, ships within 4-6 weeks. Dan: you might want to check your own webpage before replying. Cheers Darryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco, Sip, Linux, ISDN
HI Mike, 2) I could add an isdn card to the Linux box. This seems to me to be the cleanest solution, I'd make my firewall also be the asterisk server, and hopefully gain some control of tcp flows that way to more highly prioritize voice traffic +apparent simplicity, maybe fax support -s it seems most of the ISDN cards in isdn4linux are not sold in the US, the technology is stagnant, and I'm less than enthused about statements like Any CAPI based ISDN card will work when I'd prefer something like ISDN card XXX tested on an opteron running kernel X.Y.Z, using multi-link ppp and and asterisk, no problems I am using a Traverse NetJet-S card with the ISDN4Linux drivers in Asterisk on my home firewall. I was using ISDN to connect to the 'Net, but I've just -- in the last couple of months -- managed to convice Telstra let me get ADSL provisioned. I decided to keep the ISDN line for voice, so I've got a personal number and a business number coming in the ISDN. The only problem I've had is that I have had absolutely no luck in getting fax support working with the i4l driver and my questions on this list in regards to that have gone unanswered on at least 3 occasions... You can find the Traverse site at http://www.traverse.com.au/. Last time I checked they had a card for the US market (you'd need to email them to ask if the NetSpider works with I4L). HTH, Darryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RxFax problems
Hey All, I'm still (since April) having problems getting RxFax to work over an ISDN4Linux channel. Just wondering if anyone has had any luck getting it to work? I have done a CVS update today (about half hour ago) and made sure I have the latest version of spandsp according to Steve's website (spandsp-0.0.1k). When I was compiling asterisk, I got the following warnings: == gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-04/08/04-10:06:15\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fPIC -c -o app_rxfax.o app_rxfax.c In file included from ../include/spandsp.h:40, from app_rxfax.c:29: ../include/spandsp/arctan2.h: In function `arctan2': ../include/spandsp/arctan2.h:51: warning: implicit declaration of function `fabs' In file included from ../include/spandsp.h:47, from app_rxfax.c:29: ../include/spandsp/dc_restore.h: In function `fsaturate': ../include/spandsp/dc_restore.h:105: warning: implicit declaration of function `lrint' app_rxfax.c: At top level: app_rxfax.c:50: warning: no previous prototype for `t30_flush' app_rxfax.c:57: warning: no previous prototype for `phase_e_handler' gcc -shared -Xlinker -x -I/usr/src/spandsp-0.0.1/src -o app_rxfax.so app_rxfax.o -L/usr/src/spandsp-0.0.1/src -lspandsp -ltiff gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-04/08/04-10:06:15\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fPIC -c -o app_txfax.o app_txfax.c In file included from ../include/spandsp.h:40, from app_txfax.c:27: ../include/spandsp/arctan2.h: In function `arctan2': ../include/spandsp/arctan2.h:51: warning: implicit declaration of function `fabs' In file included from ../include/spandsp.h:47, from app_txfax.c:27: ../include/spandsp/dc_restore.h: In function `fsaturate': ../include/spandsp/dc_restore.h:105: warning: implicit declaration of function `lrint' app_txfax.c: At top level: app_txfax.c:46: warning: no previous prototype for `t30_flush' app_txfax.c:52: warning: no previous prototype for `phase_e_handler' gcc -shared -Xlinker -x -I/usr/src/spandsp-0.0.1/src -o app_txfax.so app_txfax.o -L/usr/src/spandsp-0.0.1/src -lspandsp -ltiff == Although it continued to compile without any other errors or warnings. When I try to receive a fax (sent from a Rockwell HCF modem) I get the following output in the asterisk console, and the originating fax doesn't handshake with it: == -- Executing Goto(Modem[i4l]/ttyI0, fax|s|1) in new stack -- Goto (fax,s,1) -- Executing Macro(Modem[i4l]/ttyI0, faxreceive) in new stack -- Executing RxFAX(Modem[i4l]/ttyI0, /var/spool/asterisk-fax/1087693402.1.tif) in new stack Jun 20 10:34:25 NOTICE[294927]: channel.c:1651 ast_set_read_format: Unable to find a path from SLINR to UNKN Jun 20 10:34:25 WARNING[294927]: app_rxfax.c:253 rxfax_exec: Unable to restore read format on 'Modem[i4l]/ttyI0' Jun 20 10:34:25 NOTICE[294927]: channel.c:1618 ast_set_write_format: Unable to find a path from UNKN to SLINR Jun 20 10:34:25 WARNING[294927]: app_rxfax.c:259 rxfax_exec: Unable to restore write format on 'Modem[i4l]/ttyI0' == Spawn extension (macro-faxreceive, s, 1) exited non-zero on 'Modem[i4l]/ttyI0' in macro 'faxreceive' == Spawn extension (fax, s, 1) exited non-zero on 'Modem[i4l]/ttyI0' -- Hungup 'Modem[i4l]/ttyI0' == Does anyone have any idea how I can get this to work? At the moment I am only really interested in receiving faxes, but sending might be nice in the future. Regards Darryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem receiving a fax with RxFAX
Hey All, I've been trying to get SpanDSP / RxFAX to work in order to set up a soft-fax machine on my asterisk system. I have asterisk CVS-04/08/04-10:06:15 and spandsp 0.0.1. This is on a Fedora Core 1 with 2.4.26 kernel. I have tried to look for a newer version of the spandsp stuff, but opencall.org does not seem to exist in DNS any more. (If it's moved location, can someone please update the Wiki - http://www.voip-info.org/wiki-Asterisk+fax ??) Anyway, I have got it compiled and installed ok. When I try to receive a fax, it does not seem to complete handshaking with the sending machine. The comment from the person sending me the test-fax was that it sounded too fast. Does anyone have any ideas? Calls are coming in via an isdn4linux interface if it would make any difference. Regards Darryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 * - * handoff
Hey All, Yeah yeah, bad form to reply to myself, but mrgoby on IRC helped me out with the answer just as I sent me question. I'm following up for the archives. Looks like there is an option in the iax.conf file called notransfer=yes. Seems to do the same thing as canreinvite=no does in sip.conf. Regards Darryl Darryl Ross wrote: Hey All, I am setting up a network of Asterisk servers using IAX2. I am wondering if it is possible to disable the handoff feature? At the moment I have 4 asterisk machines, 3 are at SOHO offices and 1 is centrally hosted in a data centre. In addition the central machine has an IAX2 link to a VOIP provider (and might be set up with more in the future). All calls are routed through that central machine. When a call is made at the moment, say from asterisk1 to asterisk2, the central machine does a native bridge of the channels (I assume this means that no codec translation needs to be done) and then gets both ends to transfer to each other. Then it releases the channels. I want to stop that occuring. I guess the SIP name for this is ReInviting. The logs look like this (passwords changed): --- -- Accepting AUTHENTICATED call from 203.221.53.223, requested format = 2, actual format = 2 -- Executing Dial([EMAIL PROTECTED]/6, IAX2/oeg_pbx:[EMAIL PROTECTED]/100,60,tT) in new stack -- Called oeg_pbx:[EMAIL PROTECTED]/100 -- Call accepted by 203.31.11.15 (format GSM) -- Format for call is GSM -- IAX2[mike]/14 stopped sounds -- IAX2[mike]/14 is ringing -- IAX2[mike]/14 stopped sounds -- IAX2[mike]/14 answered [EMAIL PROTECTED]/6 -- Attempting native bridge of [EMAIL PROTECTED]/6 and IAX2[mike]/14 -- Channel '[EMAIL PROTECTED]/6' ready to transfer -- Channel 'IAX2[mike]/14' ready to transfer -- Releasing IAX2[mike]/14 and [EMAIL PROTECTED]/6 -- Hungup 'IAX2[mike]/14' == Spawn extension (pstn-authorised, 3444100, 1) exited non-zero on '[EMAIL PROTECTED]/6' -- Hungup '[EMAIL PROTECTED]/6' --- The reason this is a problem is that the CDR records on the central machine only show a 6 second call or thereabouts, where we need it to keep track of the entire call. I've tried using the 't' and 'T' options for the Dial command, which according to the docs is supposed to make it stay in the media path, but it still hands the calls off. Any ideas?? TIA Darryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] no audio using isdn4linux channel
I'll try again using my subscribed address... Apologies to the moderators... --- Hi All, I've successfully got SIP and IAX2 working on Asterisk using X-Lite as a SIP phone and talking to a remote Asterisk over IAX2 using G729. I'm now trying to get an ISDN BRI connection working. I have a Traverse NetJet card (http://www.traverse.com.au/) which I currently have working with isdn4linux (that's how I'm connecting to the Net). I found the a message in the archives (http://www.mail-archive.com/[EMAIL PROTECTED]/msg32072.html) that says audio support is needed in the hisax driver for it to work. I have this morning downloaded and compiled 2.4.26, making sure that the audio option was turned on, but I still don't get any audio (either transmitted or received). Is there anything else I should be looking at? Eg, does it depend on the card version or perhaps a bios upgrade on the card? TIA Darryl ___ 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
