Re: [Asterisk-Users] Festival application: clipping start of sound?
IMHO the Festival application is slightly broken since it doesn't interface to the asterisk playback routines in a standard way. I've never had much luck with caching but have experienced the problem you outline on direct text conversions. This issue has been discussed on the bug tracker and this list in the past. You can hack Festival to pad out the pokayback with silence so the silence gets chopped before your sound. You can also have Festival save the sound file and then play back the sound using asterisk's standard playback routines. Both work but they're not nice solutions and add some latency, Iain --On Monday, June 14, 2004 10:58 pm +1200 Donald Gordon [EMAIL PROTECTED] wrote: Hi I'm running a bright shiny new asterisk installation, and have discovered a problem with the festival application - when it plays back the generated sound, it skips the start. If, on the other hand, it has caching turned on, then when it plays the cached sound, it doesn't skip the first word or two. I assume that this has something to do with the time taken to generate the speech - is there anything I can do about this, apart from getting a faster machine for festival? Also, files in the festival cache directory seem to be created with mode . Is there any setting I need to prod to make them readable by asterisk? I'm running the debian packaged asterisk. thanks donald ___ 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
Re: [Asterisk-Users] XML How To for Cisco 7960
--On Friday, June 11, 2004 10:46 am +0200 Stefan de Konink [EMAIL PROTECTED] wrote: http://ipphones.utelisys.net/ http://ipphones.utelisys.net/includes/cisco.inc.phps There are some perl classes on this topic too (even for image generation!). I didn't had the time to made a GD patch to use it inside PHP yet. But I hope this wil help. Anyway on Cisco.com you can find some PDF files with clear statements. Only thing that doesn't work is HTTP_PUSH :( SoftKeys don't work either :-( Iain ___ 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] XML How To for Cisco 7960
Ah, how? Which SIP version do you have - 'cos I've made innumerable tests of my own (and using Cisco code) containing SoftKey commands and the phone always barfs. Iain --On Friday, June 11, 2004 8:09 pm +1000 Simon Brown [EMAIL PROTECTED] wrote: Yes they do !! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson Sent: Friday, 11 June 2004 19:53 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] XML How To for Cisco 7960 --On Friday, June 11, 2004 10:46 am +0200 Stefan de Konink [EMAIL PROTECTED] wrote: http://ipphones.utelisys.net/ http://ipphones.utelisys.net/includes/cisco.inc.phps There are some perl classes on this topic too (even for image generation!). I didn't had the time to made a GD patch to use it inside PHP yet. But I hope this wil help. Anyway on Cisco.com you can find some PDF files with clear statements. Only thing that doesn't work is HTTP_PUSH :( SoftKeys don't work either :-( Iain ___ 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 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] XML How To for Cisco 7960
I have SIPDefault.cnf SIPxx.cnf but neither has anything in it related to XML SoftKey use nor does the manual suggest parameters related to xml use. Am I missing something? Iain --On Friday, June 11, 2004 2:48 pm +0100 Chris Bond [EMAIL PROTECTED] wrote: What you got in your sip cnf files? -Original Message- From: Iain Stevenson [mailto:[EMAIL PROTECTED] Sent: 11 June 2004 2:37 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] XML How To for Cisco 7960 Ah, how? Which SIP version do you have - 'cos I've made innumerable tests of my own (and using Cisco code) containing SoftKey commands and the phone always barfs. Iain --On Friday, June 11, 2004 8:09 pm +1000 Simon Brown [EMAIL PROTECTED] wrote: Yes they do !! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson Sent: Friday, 11 June 2004 19:53 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] XML How To for Cisco 7960 --On Friday, June 11, 2004 10:46 am +0200 Stefan de Konink [EMAIL PROTECTED] wrote: http://ipphones.utelisys.net/ http://ipphones.utelisys.net/includes/cisco.inc.phps There are some perl classes on this topic too (even for image generation!). I didn't had the time to made a GD patch to use it inside PHP yet. But I hope this wil help. Anyway on Cisco.com you can find some PDF files with clear statements. Only thing that doesn't work is HTTP_PUSH :( SoftKeys don't work either :-( Iain ___ 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 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 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] Fax via email
Maybe not. However, if the user is primarily interested in fax to email then Hylafax can do that very well. A PBX is not an essential part of a fax solution for many. Iain --On Tuesday, June 8, 2004 8:46 am +0800 Steve Underwood [EMAIL PROTECTED] wrote: Hi Iain, Your response seems to indicate that you don't know what HylaFAX and spandsp actually do :-) Regards, Steve Iain Stevenson wrote: ... might as well use hylafax. Iain --On Monday, June 7, 2004 2:15 pm +0100 Matt [EMAIL PROTECTED] wrote: Hi all. I'm looking to set up a fax via email service so that users can email a specific mailbox and receive fax's to a specific mailbox. Can this be done? I've had a look an SpanDSP and I think that's what I want but I'm not sure. Cheers Matt ___ 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
RE: [Asterisk-Users] Fax via email
.. Hylafax does that too. Iain --On Tuesday, June 8, 2004 9:15 am +0100 Matt [EMAIL PROTECTED] wrote: I'm more interested in email to fax in as much as a user could send a specifically formed email to a specific address and it be picked up and faxed out. Similarly; inbound faxes being transformed into an email. Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson Sent: 08 June 2004 09:10 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Fax via email Maybe not. However, if the user is primarily interested in fax to email then Hylafax can do that very well. A PBX is not an essential part of a fax solution for many. Iain --On Tuesday, June 8, 2004 8:46 am +0800 Steve Underwood [EMAIL PROTECTED] wrote: Hi Iain, Your response seems to indicate that you don't know what HylaFAX and spandsp actually do :-) Regards, Steve Iain Stevenson wrote: ... might as well use hylafax. Iain --On Monday, June 7, 2004 2:15 pm +0100 Matt [EMAIL PROTECTED] wrote: Hi all. I'm looking to set up a fax via email service so that users can email a specific mailbox and receive fax's to a specific mailbox. Can this be done? I've had a look an SpanDSP and I think that's what I want but I'm not sure. Cheers Matt ___ 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 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
Re: [Asterisk-Users] Fax via email
... might as well use hylafax. Iain --On Monday, June 7, 2004 2:15 pm +0100 Matt [EMAIL PROTECTED] wrote: Hi all. I'm looking to set up a fax via email service so that users can email a specific mailbox and receive fax's to a specific mailbox. Can this be done? I've had a look an SpanDSP and I think that's what I want but I'm not sure. Cheers Matt ___ 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
Re: [Asterisk-Users] Disable blind xfer
--On Friday, May 28, 2004 2:57 pm -0400 Timothy R. McKee [EMAIL PROTECTED] wrote: My SIP users need to transmit the # key as part of data entry. Asterisk intercepts and initates a transfer function. I'm almost positive I've seen this discussed somewhere, but none of my searches are finding it. This is roughly the same issue as the double hash transfer I implemented for analogue phones connecting through an ATA. Search for that. Iain ___ 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] AArgh, * and the 7960
Yes, I've read and implemented all the stuff on IAX. It's the local SIP connection and its RTP streams that's the problem. For instance I noted the strange timestamp behaviour from * on local traffic earlier. Iain --On Tuesday, May 18, 2004 1:56 pm -0600 Rich Adamson [EMAIL PROTECTED] wrote: I've just had the most appalling performance from * ever. Dialling: Cisco 7960 = asterisk = IAX produces sound drop outs so extreme that the call is useless. I noted this in an earlier post. Dialling: Cisco ATA186 = asterisk = IAX is fine. Frankly, I think this is such a bad problem that it should be sorted in advance of any of the new features that seem to be getting such prominence nowadays. It was not present earlier in the year and I haven't upgraded my 7960. So I don't think you can point the finger entirely in Cisco's direction. The problem has been discussed multiple times over the last several weeks. To recap, there is two things needed to incure the problem: 1. cisco 7960 phone (it discards packets with uneven timestamps) 2. asterisk had an iax problem that was fixed about a month ago assoicated with uneven timestamps. The distant iax system will need to be upgraded to fairly recent code. See previous posts for more detail. ___ 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
RE: [Asterisk-Users] AArgh, * and the 7960
--On Tuesday, May 18, 2004 1:42 pm -0500 Nik Martin [EMAIL PROTECTED] wrote: Out of context, this isn't much information. Is your network connection OK? Yes, AFAIK - I'm running all the traffic shaping / prioritisation stuff mentioned on the list Is your broadband provider having troubles? AFAIK - but then it is BT Openworld ;-) Has some upstream hardware changed that you may not be aware of? My call is going through IAXTEL so Digium must know if there's a problem. A test IVR system within IAXTEL would be nice for testing. Iain ___ 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] AArgh, * and the 7960
--On Tuesday, May 18, 2004 1:43 pm -0500 brian [EMAIL PROTECTED] wrote: Strange I do 7960 = * = IAX all day long without one jitter or any bad audio. Now if both ends are NOT running the very latest(within the last month or so) CVS-head for example if you have say a 2 month old chan_iax2.c on one end then oh boy you're in for a bad time they need to update. Is the 7960 using SIP? The problem happens with the latest * (cvs co asterisk). I think it's quite likely the local RTP handling that's the problem. Iain ___ 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] AArgh, * and the 7960
I have ethereal installed and I'll do a full call trace. The Catch 22 is I
don't have access to access to a source of repeatable (ie recorded) content
accessed through IAX. That would help in producing traces for the ATA and
7960 for comparison. I mainly use IAX for non-critical international
business calls to people who wouldn't want to be * testers.
Iain
--On Tuesday, May 18, 2004 7:22 pm -0600 brian k. west [EMAIL PROTECTED]
wrote:
Lets look at this and FIX the problem instead of hacking it. What you
need to do is install etherreal and capture a call and parse the
timestamp info to see if they are slipping. Because they are perfect
here.
bkw
- Original Message -
From: Brian Cuthie [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, May 18, 2004 5:07 PM
Subject: Re: [Asterisk-Users] AArgh, * and the 7960
Iain,
This is a known issue with the Cisco phone and Asterisk having to do
with a change made later in the cvs tree. Try 1.0 stable, or modify
rtp.c to comment out the two lines as follows:
/* Re-calculate last TS */
rtp-lastts = rtp-lastts + ms * 8;
// if (!f-delivery.tv_sec !f-delivery.tv_usec) {
/* If this isn't an absolute delivery time,
Check if it is close to our prediction,
and if so, go with our prediction */
if (abs(rtp-lastts - pred) 640)
rtp-lastts = pred;
else {
ast_log(LOG_DEBUG, Difference is %d, ms
is %d\n, abs(rtp-lastts - pred), ms);
mark = 1;
}
// }
} else {
This seems to work for me. Others may have more insight.
-brian
Nik Martin wrote:
Out of context, this isn't much information. Is your network
connection
OK?
Is your broadband provider having troubles? Has some upstream hardware
changed that you may not be aware of?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Iain Stevenson
Sent: Tuesday, May 18, 2004 1:29 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] AArgh, * and the 7960
I've just had the most appalling performance from * ever. Dialling:
Cisco 7960 = asterisk = IAX
produces sound drop outs so extreme that the call is useless.
I noted this
in an earlier post. Dialling:
Cisco ATA186 = asterisk = IAX
is fine.
Frankly, I think this is such a bad problem that it should be
sorted in
advance of any of the new features that seem to be getting
such prominence
nowadays. It was not present earlier in the year and I
haven't upgraded my
7960. So I don't think you can point the finger entirely in Cisco's
direction.
Iain
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Re: [Asterisk-Users] AGI/php script not working
'SAY NUMBER 123 #* isn't a valid * command - it should be saynumber followed by a valid string of arguments. Do a show application saynumber in *. Iain --On Thursday, May 20, 2004 7:14 am -0400 Jer [EMAIL PROTECTED] wrote: Dear all I am just getting started with AGI so I wrote the following script as a simple test but all that happens is silence before it times out and hangs up can someone help to get me started? yet if i use the agi-test.agi script everything works I don't see the difference Thanks php -q ?php fputs(STDOUT 'SAY NUMBER 123 #*\n'); $lin = fgets(STDIN); ? yet all I get on the console is -- Launched AGI Script /var/lib/asterisk/agi-bin/test.php -- AGI Script test.php completed, returning 0 my conf file looks like exten = 4000,1,Wait,1 ; Wait exten = 4000,2,Answer ; Answer exten = 4000,3,AGI,test.php ; run script ___ 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
Re: [Asterisk-Users] AGI/php script not working
OK, but I have AGI working and you don't - so please allow me the error since it's a while since I worked on this, Of course, it would help if * used consistent syntax for identical commands in extensions.conf and AGI, but that's another debate. Why not check the logs for php and * and post anything relevant here. Enable the maximum debugging support in *. Iain --On Thursday, May 20, 2004 2:44 pm +0300 Apollon Koutlides [EMAIL PROTECTED] wrote: Iain Stevenson wrote: 'SAY NUMBER 123 #* isn't a valid * command - it should be saynumber followed by a valid string of arguments. Do a show application saynumber in *. In the meantime, you might as well try a show agi yourself :-) Apollon Koutlides ___ 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
Re: [Asterisk-Users] Re: Grandstream v1.0.4.68 firmware
--On Tuesday, May 18, 2004 12:30 pm -0400 Stephen R. Besch [EMAIL PROTECTED] wrote: P.S. Grandstream, if you are listening, then Early Dial is still broken! It's been many months now to fix what apparently is just a counter bug. Come on, let's get this fixed. Here, here! Iain ___ 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] AArgh, * and the 7960
I've just had the most appalling performance from * ever. Dialling: Cisco 7960 = asterisk = IAX produces sound drop outs so extreme that the call is useless. I noted this in an earlier post. Dialling: Cisco ATA186 = asterisk = IAX is fine. Frankly, I think this is such a bad problem that it should be sorted in advance of any of the new features that seem to be getting such prominence nowadays. It was not present earlier in the year and I haven't upgraded my 7960. So I don't think you can point the finger entirely in Cisco's direction. Iain ___ 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 and Cisco 7960 problems persist (for me, anyway)
This isn't really the issue. Up until a week ago or so everything worked fine with a hallf duplex hub. Now it doesn't - so I suspect some code change made in * is responsible. I think * must maintain backwards compatibility with existing hardware or many people will get fed up with constant degradation of sound quality. Iain --On Friday, May 07, 2004 14:15:47 -0600 James Sizemore [EMAIL PROTECTED] wrote: I checked-out CVS Head today to get realm support, I have over hundred Cisco phone on my servers and I have not noticed any Qos problems. You may want to check the duplex of your switches and Asterisk boxes. If you don't have full duplex, that is more then likely your problem. Brian Cuthie wrote: It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box using a TDM400P interface). Cheers, Brian ___ 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 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 and Cisco 7960 problems persist (for me, anyway)
I've had this too, reported it as a bug last week and got my butt kicked for not being responsive enough in providing support to sort it out. You could file another bug report but be sure to have a thick book ready to stuff down your trousers. Iain --On Friday, May 7, 2004 10:43 am -0400 Brian Cuthie [EMAIL PROTECTED] wrote: It seems that each time I get a new checkout of * from CVS my Cisco 7960 works worse than before. I know this stuff's in flux, so I mention this in case it's news. Anyone else having trouble? What I'm seeing (er, hearing) is really choppy audio. The previous version I had installed had fairly frequent audio dropouts (not present when I make the same calls through the same * box using a TDM400P interface). Cheers, Brian ___ 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
Re: [Asterisk-Users] X100P Answer
You've probably got callerID enabled in zapata.conf. That will cause a wait of several rings whilst * looks for the caller ID info. Since this only works in the US (or pkaces with similar phone systems), disabling it in other territories saves the ring delay. Make sure you have this in zapata.conf usecallerid=no IAin --On Friday, April 23, 2004 2:55 pm +0100 Mark Olliver [EMAIL PROTECTED] wrote: Hi, I seam to have a problem working out how to get my X100P to answer after 1 ring. Currently it is working fine and connects to the switchboard menu correctly but just does it after 4 rings, which I would prefer if we could reduce. Thanks Mark -- Mark Olliver Thermeon Europe Ltd. e-Card: http://www.thermeoneurope.com/e-Card/mpo Email [EMAIL PROTECTED] Web www.thermeoneurope.com Support 0906 515 0908 Int. Support +44 1293 864 341 Support Email [EMAIL PROTECTED] Sales +44 1293 864 334 Sales Email [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] Missing vm feature - turn off voicemail
Listening to the options on the voicemail system it seems to be missing a feature for users to turn voicemail off completely. This seems a rather glaring omission. Does the feature of turning off message recording via the phone exist - or does it need a patch? Iain ___ 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] Re: Analogue telephone cards for the UK
--On Saturday, April 10, 2004 10:42:26 +0100 Paul Tyreman [EMAIL PROTECTED] wrote: Thanks for all the replies. Can someone tell me if it is possible to put two of these X100P cards into the same machine to order to gain access to two BT landlines ? I believe so although problems have been reported with certain motherboards - best to search this list before buying. Would it also be possible for someone to outline in a bit more detail the procdue for limiting which phones have access via the card as I am new to Asterisk. You need to define a context for outgoing calls which will include dial commands for the X100P. You then define additional contexts for local phones. Only those local contexts that include the outgoing context will be able to make outgoing calls. Start with a bare bones extensions.conf or you'll find * very hard going. What happens when someone calls the number of the line the card is on - Do all phones ring or what happens ? You define that in extensions.conf. Incoming calls will land in the context you specify in /etc/asterisk/zapata.conf Is that auto attendant thing a real possiblity. What I would idealy like is this... Welcome. If you know the extention you wish to call, press * now and then dial it. Otherwise, press 1 for Family A, 2 for Family B and 3 for Family C. If the user Presses 1, Press 1 for Person A, Press 2 for Person B. etc ? Is that possible ? ... I dunno - sorry Iain ___ 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] Re: Analogue telephone cards for the UK
--On Saturday, April 10, 2004 11:55:26 +0100 Paul Tyreman [EMAIL PROTECTED] wrote: What I want to do is have the asterisk server sat in my house and used by my family to access the BT landline and to recieve calls made to that landline. If it is not possible to do the auto attendant thing then so be it, I will just have all phones in my house ring when a call is made on the BT line. That should be easy, right ? .. yes, this is what I have for my SOHO setup. In addition to running the server just for my house, I want to have other memebers of my extended family link up to the server via their broadband connections so we can make free calls to each other over the internet connections. ... looks like a case for SIP or IAX clients. What I don't want is for other members of my family (who are not resident in my house) to be able to make calls on my BT landline, but I do want them to be able to make unlimited calls to other extentions on the asterisk server. ... they need to be in a context that has no access to Dial commands that target the X100P. Since I already pay monthly for broadband, I am not very keen to start paying more for an IDSN line which will only be used for this project. I don't use / need caller ID on external calls, so thats not an issue. Does that all make sence ? yes - first choose the type of client your extended family will use. Then create an appropriate iax.conf or sip.conf and include entries for all authorised users. Create contexts for 'local' and 'extended-family users in extensions.conf. Assign users to either context as appropriate in sip.conf or iax.conf. Assign extensions numbers, set up Dial commands etc ... Most is explained in the asterisk manual or on the VoIP wiki. Iain Thanks, Paul. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Posted At: 10 April 2004 11:16 Posted To: Asterisk-Users Conversation: [Asterisk-Users] Re: Analogue telephone cards for the UK Subject: Re: [Asterisk-Users] Re: Analogue telephone cards for the UK It sounds like you are trying to share the PBX between multiple people.. I would suggest getting an ISDN BRI line and an AVM Fritz card (using the chan_capi driver).. This will give you two lines onto which you can get 8 MSN's (an MSN is another number coming in on the same BRI).. You can setup Asterisk to route the calls to the correct phones or group of phones based on the number that was called.. If you are in the UK there are plenty of Fritz cards around and this method will also allow you to have CallerID if you want it where the analog cards have issues with CallerID.. Later.. ___ 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] Re: Analogue telephone cards for the UK
--On Saturday, April 10, 2004 17:47:24 +0100 Paul Tyreman [EMAIL PROTECTED] wrote: Sorry to sound stupid, but where can I get copied of the Asterisk manual ? http://www.asterisk.org/index.php?menu=support#handbook_project What is the VoIP wiki and where can I get that too ? The wiki is a searchable site with lots on asterisk - http://www.voip-info.org/tiki-index.php Iain ___ 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] Newbie....
--On Wednesday, March 31, 2004 2:00 pm -0500 Hall, Eric M. [EMAIL PROTECTED] wrote: I have a question for the group. To get this running do I need any Digium Cards? I understand I will need them to connect to the public phone system. I'm looking at just using IP Phones or IP Softphones just to test this app. Most asterisk functions will work without a Digium zaptel card - so try it out! Iain ___ 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] Opinion poll: best SIP phones for asterisk?
--On Monday, March 29, 2004 8:24 am -0500 Kevin [EMAIL PROTECTED] wrote: Hi All- As I'm doing this, I'm considering installing an asterisk box at my office (about 6-10 different phone stations) and would like to get opinions on the best quality and/or most well-supported SIP hard phones and SIP soft phone clients. Lot's of references to this topic on the list - a quick search will provide loads of feedback. At a top level: - cheap option - Grandstream Budgetone - works well but current firmware is buggy - best option - Cisco - good * integration. Make sure you get the phone supplied with SIP installed to avoid any support issues I see from John Todd's config files that the ATA-186 has (or had) some problems with asterisk, and I'm guessing that there are probably some other flukes that crop up with certain phones being connected to asterisk, so I thought I'd ask here before purchasing something in hopes of avoiding unnecessary hassles. - works fine but call transfers can be tricky to action Iain ___ 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] Opinion poll: best SIP phones for asterisk?
--On Monday, March 29, 2004 2:09 pm + Hermann Wecke [EMAIL PROTECTED] wrote: Which one? I'm running one the latest image available at http://www.grandstream.com/BETATEST/ (b14p4.54.zip) and my * and my GS are working OK. The 4.53 was buggy, but I can't find a problem (so far) with 4.54 I'm running 4.50 because of adverse reports of 4.53 etc. Is abbreviated dialling (aka Early Dial) working yet - it's been out of commission for most firmware from 35 - 50 releases. Iain ___ 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 7960 SIP Images
Welcome to the very much less than wonderful world of Cisco software support. When will those guys simply make the software downloadable straight away from their website for a modest fee? Iain --On Saturday, March 27, 2004 1:43 am -0600 Mitchell S. Sharp [EMAIL PROTECTED] wrote: I just received my first Cisco 7960 today and was looking forward to playing with it this weekend, however I can't seem to get it working via skinny (can't find any information via the wiki regarding what needs to be on the tftp server for skinny). I would like to get my hands on the SIP images to play with it. I know I have to get a support contract through Cisco to get download access via their site which you can bet I'm going to do Monday morning, but I was hoping to work with it this weekend while I have the time. I found the release 4.4 SIP image, but it won't take due to a bug that was evidently fixed around v3.? (4k tftp buffer, and the new image is larger). At least I have a really expensive pretty phone sitting on my desk now! :-) Mitch Sharp ___ 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
RE: [Asterisk-Users] Cisco 7960 SIP Images
.. not sure this applies outside the US - or I'd reach for the credit card. Iain --On Friday, March 26, 2004 11:29 am -0500 Chris HARIGA [EMAIL PROTECTED] wrote: If you pay 8 USD for 1 year support you can download the image :) Best regards, Chris HARIGA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson Sent: Saturday, March 27, 2004 4:06 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 SIP Images Welcome to the very much less than wonderful world of Cisco software support. When will those guys simply make the software downloadable straight away from their website for a modest fee? Iain --On Saturday, March 27, 2004 1:43 am -0600 Mitchell S. Sharp [EMAIL PROTECTED] wrote: I just received my first Cisco 7960 today and was looking forward to playing with it this weekend, however I can't seem to get it working via skinny (can't find any information via the wiki regarding what needs to be on the tftp server for skinny). I would like to get my hands on the SIP images to play with it. I know I have to get a support contract through Cisco to get download access via their site which you can bet I'm going to do Monday morning, but I was hoping to work with it this weekend while I have the time. I found the release 4.4 SIP image, but it won't take due to a bug that was evidently fixed around v3.? (4k tftp buffer, and the new image is larger). At least I have a really expensive pretty phone sitting on my desk now! :-) Mitch Sharp ___ 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 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
RE: [Asterisk-Users] Cisco 7960 SIP Images
--On Saturday, March 27, 2004 4:52 pm -0500 Ray Burkholder [EMAIL PROTECTED] wrote: Iain Stevenson wrote: .. not sure this applies outside the US - or I'd reach for the credit card. Iain --On Friday, March 26, 2004 11:29 am -0500 Chris HARIGA [EMAIL PROTECTED] wrote: If you pay 8 USD for 1 year support you can download the image :) Best regards, Chris HARIGA No, you can't use a credit card. You have to send the #$!@@$#'s a check. It's really stupid, but it's the Cisco way. John Or purchase a Smartnet from your local Cisco reseller. Unfortunately I haven't found any reseller offering cheap (or Smartnet) contracts in the UK. There always seems to be a steep premium. Iain ___ 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 as an IETF Standard?
--On Wednesday, March 24, 2004 11:13 am -0600 Steven Sokol [EMAIL PROTECTED] wrote: I have seen a number of postings cross this list that mention the possibility of standards-tracking IAX2 with the IETF (generating an RFC, etc.). Has that gone anywhere? What would it take to make it happen? I think there are significant marketing advantages to generating an Informational RFC for IAX2. The fact that IAX does cross firewalls is very important in the consumer market and of course helpful for everyone else. At the moment Sk(h)ype gain significant PR mileage from this point. Most of the Press and Analyst community seem to leave their critical faculties turned off when Skype is mentioned relating only the good points and not the bad (security and bandwidth issues for end users, scalability etc). Showing that there is a credible and standard alternative approach seems to me to be a very good idea. Iain ___ 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] X100P fails to detect user hung up
I think this has been discussed a lot in the last 3 days - do some legwork before posting! Iain --On Wednesday, March 24, 2004 3:53 pm -0800 Ron McMillin [EMAIL PROTECTED] wrote: I am using the wildcard X100P with *. PSTN line comes in to the FXO port of this card. Everything works fine most of the time. However, occasionally Asterisk doesn't seem to be able to detect the user has hung up and therefore tie up the line for quite a long time. Does anyone know if there's anything I can do to fix this problem? thanks Ron ___ 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] UK PSTN and x100p
--On Sunday, March 21, 2004 8:11 pm + Dee Lowndes [EMAIL PROTECTED] wrote: If I find the voltage drop out can I configure the x100p to do it based on the new voltage drop. If so where and how? To a certain extent yes. Im fact, in the absence of measurements you could just try a couple of things in the code. You need to edit wcfxo.c and then recompile and reload the wcfxo and zaptel modules. In there you will find a couple of options - one for Japanese networks and one for zero battery ring. Try enabling these and see if they help. I wrote the 'xero battery ring' patch to cope with a hangup problem on my ISDN TA. It may or may not help with your Telewest issues. Iain ___ 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: cpu load 99%
--On Monday, March 22, 2004 12:51 pm +0100 Matteo Rancilio [EMAIL PROTECTED] wrote: You're right :) I'm using Asterisk 7.2 on a SuSE 8.2 installation. Hardware: Dual Intel PIII 1Gb ram AVM Fritz! ISDN card SIP CISCO Phones Codec g711 (switching today to g729) ... and what applications? AGI, Festival? Festival completely stuffed my * server once. Iain ___ 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] Using the pound (#) key while in a call
I assume you're using Dial with the Tt options to enable transfer? If you need to keep the transfer you may need something like the double hash patch I posted last week. Iain --On Friday, March 19, 2004 1:39 am -0600 Mitchell S. Sharp [EMAIL PROTECTED] wrote: Haven't been able to find anything on this the past couple days, so I'm asking ya'll! When I call my bank or a vendor or who ever through the * server and they want me to enter a pin number followed by the pound sign, what's the trick? Of course when I hit pound, * asks me where I want to transfer the call. The only solution I can think of is to not allow my 800 dial outs the ability to transfer. Any help is greatly appreciated! Mitch Sharp Innovative Solutions ___ 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] Using the pound (#) key while in a call
It went to the list 'cos Mark's not in favoyr of the patch and someone wanted it urgently - search the list on my name and you'll find it. Iain --On Friday, March 19, 2004 5:01 am -0600 Matthew Marlowe [EMAIL PROTECTED] wrote: I assume when you say posted, you meant on bugs.digium.com? I cant find it, can you give me the ID please. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson Sent: Friday, March 19, 2004 4:00 AM To: Asterisk Users Subject: Re: [Asterisk-Users] Using the pound (#) key while in a call I assume you're using Dial with the Tt options to enable transfer? If you need to keep the transfer you may need something like the double hash patch I posted last week. Iain --On Friday, March 19, 2004 1:39 am -0600 Mitchell S. Sharp [EMAIL PROTECTED] wrote: Haven't been able to find anything on this the past couple days, so I'm asking ya'll! When I call my bank or a vendor or who ever through the * server and they want me to enter a pin number followed by the pound sign, what's the trick? Of course when I hit pound, * asks me where I want to transfer the call. The only solution I can think of is to not allow my 800 dial outs the ability to transfer. Any help is greatly appreciated! Mitch Sharp Innovative Solutions ___ 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 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] Using the pound (#) key while in a call
... then it's not working and you need the patch from the list. Iain --On Friday, March 19, 2004 8:04 am -0600 Matthew Marlowe [EMAIL PROTECTED] wrote: I found it on the bugs site actually.. It's weird though, patch applied successfully although hitting one # still goes right into the transfer. A little odd I'd say. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson Sent: Friday, March 19, 2004 7:19 AM To: Asterisk Users Subject: RE: [Asterisk-Users] Using the pound (#) key while in a call It went to the list 'cos Mark's not in favoyr of the patch and someone wanted it urgently - search the list on my name and you'll find it. Iain --On Friday, March 19, 2004 5:01 am -0600 Matthew Marlowe [EMAIL PROTECTED] wrote: I assume when you say posted, you meant on bugs.digium.com? I cant find it, can you give me the ID please. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson Sent: Friday, March 19, 2004 4:00 AM To: Asterisk Users Subject: Re: [Asterisk-Users] Using the pound (#) key while in a call I assume you're using Dial with the Tt options to enable transfer? If you need to keep the transfer you may need something like the double hash patch I posted last week. Iain --On Friday, March 19, 2004 1:39 am -0600 Mitchell S. Sharp [EMAIL PROTECTED] wrote: Haven't been able to find anything on this the past couple days, so I'm asking ya'll! When I call my bank or a vendor or who ever through the * server and they want me to enter a pin number followed by the pound sign, what's the trick? Of course when I hit pound, * asks me where I want to transfer the call. The only solution I can think of is to not allow my 800 dial outs the ability to transfer. Any help is greatly appreciated! Mitch Sharp Innovative Solutions ___ 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 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 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] MOH: Copyright issues?
I guess that means every * server needs to play old Victorian Music Hall favourites: Bicycle Built For Two Daddy WouldnÂ’t Buy Me A Bow Wow Hello, Hello, WhoÂ’s Your Lady Friend? The Man on the Flying Trapeze ... and many more Iain --On Friday, March 19, 2004 12:59 pm -0800 George Pajari [EMAIL PROTECTED] wrote: See also: http://www.bmi.com/licensing/business/groupb/faq/musiconhold_questions.asp http://www.socan.ca/jsp/en/resources/tariffs.jsp (see category 15B) ___ 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
Re: [Asterisk-Users] Festival
Look here ... http://www.cstr.ed.ac.uk/projects/festival/ Iain --On Friday, March 19, 2004 4:10 pm -0600 Justin Carlson [EMAIL PROTECTED] wrote: I am sorry if this is a silly question but I can not seem to locate the festival binaries. does this come with asterisk or is it another project? ___ 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
Re: [Asterisk-Users] Monastery Devel snapshot
... just installed this. The database updates OK but status.php shows no active channels (either SIP to SIP or SIP to voicemail). Iain --On Thursday, March 18, 2004 11:51 am -0500 Tim Sailer [EMAIL PROTECTED] wrote: I just pushed out a snapshot of the -devel version of monastery. ftp://ftp.buoy.com/pub/asterisk/monastery-devel.tgz Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 IAX 17003992910 ___ 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
Re: [Asterisk-Users] Monastery Devel snapshot
I'll answer my own question ... If you don't call the database asterisl you need to edit in the name you do use to status.php otherwise monastery behaves as though nothing is happening rather than flagging an error ;-) Iain --On Thursday, March 18, 2004 5:51 pm + Iain Stevenson [EMAIL PROTECTED] wrote: ... just installed this. The database updates OK but status.php shows no active channels (either SIP to SIP or SIP to voicemail). Iain --On Thursday, March 18, 2004 11:51 am -0500 Tim Sailer [EMAIL PROTECTED] wrote: I just pushed out a snapshot of the -devel version of monastery. ftp://ftp.buoy.com/pub/asterisk/monastery-devel.tgz Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 IAX 17003992910 ___ 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 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] Hangup X100P Issues
What sort of phone line are you using? Connecting an X100P to a PBX line or ISDN TA can cause the problems you mention. Iain --On Wednesday, March 17, 2004 7:37 am -0600 [EMAIL PROTECTED] wrote: Hullo! It appears that the X100P (FXO) does somehow not passes the 'hangup' signaling *. Sample Scenario 1: I call in on external line X100P. I successfully ring an extension. The extension answers. [we have an established call going on now] I hangup (from the external call). Listening to the extension, I just hear a faitn click and then *silence* as if the caller stopped talking. Eventually the person on the extension will actually hangup, releasing the FXO Sample Scenario 2: As above, I call in through the X100P. I dial an extension for VoiceMailMain. Somewhere in the process, I just hangup. The VoiceMailMain keeps happily looping until *eventually* it actually times out. I have tried both scenarions dialing in through IAX (VoicePulse), and both work as expected: i.e. caller hangs up, callee (on extension) hears a 'click' followed by 'congestion' tone. The 'hangup' event is detected. I searched the archives, but could not find a solution. Any ideas, TIA Willy Willy Wouters ypOne Publishing ___ 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
Re: [Asterisk-Users] XML Phone book software.
It's quite easy to write an LDAP interface. There are code snippets on the web and I can send you my very quick hack, if you like. Iain --On Thursday, March 11, 2004 4:06 pm -0600 Brian R. Swan [EMAIL PROTECTED] wrote: Hi gang, I'm looking into writing a some phone book XML/PHP software for my Cisco phones. Specifically, I'd like to be able to use a web interface (on the computer) to maintain a contact list, and then dial from it on the phone. Maybe using MySql on the back end or something (to be determined). Before I start, and duplicate something else that exists, I wanted to see if anyone has heard of software like that? Searches of Sourceforge, Freshmeat, and Google didn't turn up much or anything. Thanks! Brian ___ 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
Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.
I hacked the Wait command to wait in increments of 100ms. The 7960 needs about 300ms delay after answer to play the sound properly. ATA186's work fine without any delay for me. A finer grained 'Wait' would be helpful in developing workarounds for this sort of problem. Iain --On Wednesday, March 10, 2004 6:04 pm -0800 Andrew Gillham [EMAIL PROTECTED] wrote: Steve Creel wrote: On Wed, 10 Mar 2004, John Fraizer wrote: For what it's worth, I don't have any delay between answer and audio with my asterisk server and 7960G either originating or answering. It doesn't matter if it's a call to/from another SIP/IAX device or to/from PSTN. It's pretty much instant (not detectable by humans at least). So, there may be some truth to the fact that the delay is caused by the Asterisk install in your case. There are so many variables that it is very hard to tell but, since I don't see the delay, I am leaning towards it being an Asterisk implementation issue. Can you test this with an extension that goes into VoiceMailMain(). My 7960 and 7960G phones both get the first couple letters of Commedian Mail cut off (usually ...median Mail). Just trying to quantify the delay we're talking about... exten = 6500,1,Answer exten = 6500,2,Wait,1 exten = 6500,3,VoicemailMain2 Or should I say, Me too! Is this the bug for the case in question? CSCed48311: Media takes 0.4 sec to be set up Thanks. -Andrew ___ 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
Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring.
--On Thursday, March 11, 2004 3:17 am -0500 James Golovich [EMAIL PROTECTED] wrote: As of 3/4/2004 in cvs head and stable the Wait application has accepted time with fractions of a second. So 0.1 would be 100ms, 0.3 would be 300ms, etc. James Thanks, that makes a workaround for the 7960 problem this: exten = 40,1,Answer exten = 40,2,Wait,0.3 exten = 40,3,VoicemailMain2 Iain ___ 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] Outbound Transfer and the # key
Try the attached patch. Go to your asterisk root directory and type: patch -p0 path_to_patch/Parking.patch .. then rebuild asterisk. Iain --On Wednesday, March 10, 2004 7:43 am -0500 John Congdon [EMAIL PROTECTED] wrote: I have applied the patch and restarted Asterisk. But it still only requires a single # to transfer. Did I possibly miss something? This is just to show that it was applied. [EMAIL PROTECTED] asterisk]# pwd /usr/src/asterisk [EMAIL PROTECTED] asterisk]# patch -p0 ../old_asterisk/doublehash.patch patching file res/res_parking.c Reversed (or previously applied) patch detected! Assume -R? [n] Apply anyway? [n] Skipping patch. 3 out of 3 hunks ignored -- saving rejects to file res/res_parking.c.rej John On Mar 9, 2004, at 4:53 PM, mattf wrote: There is a better way to deal with this, it's the doublehash patch. This patch makes it so you have to press the hash key twice to transfer a call instead of once as is default in Asterisk. Sad thing is that every time the parking code changes the patch has to change(sometimes twice a week) and I don't have a patch for the most recent CVS. I've asked numerous times for some wonderful Asterisk-code-God(please Mark ;)) to make it a configurable variable in the parking.conf file but noone seems to think it's worthy of doing. It's actually a rather simple code change from what I can guess reading the patch code. I've been told that the core developers(Mark) don't want to mess with doublehash, but maybe if enough people say they want it we can get them to make this harmless addition to the parking code. Here's a bug where it's been talked about: http://bugs.digium.com/bug_view_page.php?bug_id=885 MATT--- -Original Message- From: John Congdon [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 09, 2004 4:24 PM To: Asterisk Mailling List Subject: [Asterisk-Users] Outbound Transfer and the # key Has there been any resolution to this? Does anyone have a good way to allow someone to choose whether they want to be able to transfer a call vs send the # to the other end. Is there a simple way to change the Transfer key for # to *? John ___ 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 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 Parking.patch Description: Binary data
RE: [Asterisk-Users] Outbound Transfer and the # key
Oh dear. You can either manually enter in the missing line or apply the attached patch as before (you need a clean res_parking.c which you can get by deleting the file and then doing cvs co asterisk again). This patch works on my system updated to the latest cvs. Iain --On Wednesday, March 10, 2004 4:54 pm -0500 mattf [EMAIL PROTECTED] wrote: Here's my patch results: [EMAIL PROTECTED] asterisk]# patch -p0 ./Parking.patch patching file res/res_parking.c Hunk #1 FAILED at 25. Hunk #2 succeeded at 228 (offset 13 lines). Hunk #3 succeeded at 288 (offset 12 lines). Hunk #4 succeeded at 408 (offset 13 lines). 1 out of 4 hunks FAILED -- saving rejects to file res/res_parking.c.rej [EMAIL PROTECTED] asterisk]# cat res/res_parking.c.rej *** *** 25,30 #include asterisk/musiconhold.h #include asterisk/config.h #include asterisk/cli.h #include stdlib.h #include errno.h #include unistd.h --- 25,31 #include asterisk/musiconhold.h #include asterisk/config.h #include asterisk/cli.h + #include asterisk/indications.h #include stdlib.h #include errno.h #include unistd.h is the first fail a bad thing? This is CVS from 15 minutes ago. Thanks, MATT--- -Original Message- From: Iain Stevenson [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 10, 2004 4:33 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Outbound Transfer and the # key Try the attached patch. Go to your asterisk root directory and type: patch -p0 path_to_patch/Parking.patch .. then rebuild asterisk. Iain --On Wednesday, March 10, 2004 7:43 am -0500 John Congdon [EMAIL PROTECTED] wrote: I have applied the patch and restarted Asterisk. But it still only requires a single # to transfer. Did I possibly miss something? This is just to show that it was applied. [EMAIL PROTECTED] asterisk]# pwd /usr/src/asterisk [EMAIL PROTECTED] asterisk]# patch -p0 ../old_asterisk/doublehash.patch patching file res/res_parking.c Reversed (or previously applied) patch detected! Assume -R? [n] Apply anyway? [n] Skipping patch. 3 out of 3 hunks ignored -- saving rejects to file res/res_parking.c.rej John On Mar 9, 2004, at 4:53 PM, mattf wrote: There is a better way to deal with this, it's the doublehash patch. This patch makes it so you have to press the hash key twice to transfer a call instead of once as is default in Asterisk. Sad thing is that every time the parking code changes the patch has to change(sometimes twice a week) and I don't have a patch for the most recent CVS. I've asked numerous times for some wonderful Asterisk-code-God(please Mark ;)) to make it a configurable variable in the parking.conf file but noone seems to think it's worthy of doing. It's actually a rather simple code change from what I can guess reading the patch code. I've been told that the core developers(Mark) don't want to mess with doublehash, but maybe if enough people say they want it we can get them to make this harmless addition to the parking code. Here's a bug where it's been talked about: http://bugs.digium.com/bug_view_page.php?bug_id=885 MATT--- -Original Message- From: John Congdon [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 09, 2004 4:24 PM To: Asterisk Mailling List Subject: [Asterisk-Users] Outbound Transfer and the # key Has there been any resolution to this? Does anyone have a good way to allow someone to choose whether they want to be able to transfer a call vs send the # to the other end. Is there a simple way to change the Transfer key for # to *? John ___ 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 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 Parking.patch Description: Binary data
Re: [Asterisk-Users] x100p Q.
The root cause of the problem is the 3 wire phone wiring in the UK compared to the 2 wire wiring in the US. I've had the problem you mention just using ordinary phones! I suspect that a socket somewhere has been wired up with wires crossed. Your X100P probably needs to go straight across the incoming line. Iain --On Saturday, March 6, 2004 10:03 am + Jon Lawrence [EMAIL PROTECTED] wrote: Hi everyone. I've now managed to my basic voip setup working, but I have a problem with my fxo cards. If I plug the cards into the pstn line whilst a normall phone is also plugged in, the normal phone continually rings. I'm convinced that this is a problem with the wiring but I don't know what/why. The * box works perfectly (with the exception of the callerid) so long as I don't have another phone plugged in. I can't just unplug all the other phones - the sky box + alarm system must remain plugged in. I can still ring out on the other phones and also on the * box, but the constant ringing is obviously a problem :) Is this normal ? Has anyone else seen this ? fyi I'm based in the UK. TIA Jon Lawrence ___ 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] FWD registration faillures
Anyone else seeing SIP registration requests rejected by FWD? I don't seem to be able to register any longer - even though my SIP config remains the same. Iain ___ 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] Fax detected, but no fax extension
Edit the top level * Makefile to enable this: OLD_DSP_ROUTINES then rebuild and reinstall * Iain --On Monday, March 1, 2004 7:09 pm -0300 listas iPfone [EMAIL PROTECTED] wrote: Hi! Every time i make or receive a call with my x100p i receive that notice: NOTICE[1234379840]: chan_zap.c:3640 zt_read: Fax detected, but no fax extension Maybe that is problem with brazilian lines? How can i stop it? Miklos iPFONE Telefonia IP Rua Caio Graco 735 São Paulo SP iPBX +55 11 3801-3702 FWD 64662 sip:[EMAIL PROTECTED] www.ipfone.com.br [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Message waiting light not coming on
Works perfectly fine for me - but I'm not using rfc2683 - my Grandstream uses the latest firmware and SIP INFO. Iain --On Thursday, February 26, 2004 12:55 am -0500 Rana Dutt [EMAIL PROTECTED] wrote: I cannot get the Message Waiting Light (MWL) on my Grandstream phone to turn on when I leave a new voice mail message for that phone. I have specified the correct mailbox in my sip.conf as follows: [200] type=friend username=200 host=dynamic context=dialout callerid=200 dtmfmode=rfc2833 mailbox=200 I also have an IpDialog Siptone II phone, and can't get the MWL to work on that either. Did anyone have a problem like this? -Ron ___ 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
Re: [Asterisk-Users] DSL (DMT) goes down when X100 plugged in
I'd reach for the Oxometer on that one - 36k shouldn't make any difference. However, the X100P may be introducing some capacitance on the line that would affect the ADSL signals - but the purpose of filters is to stop this problem. Maybe it's worth trying another filter between the X100P and your other phones. Iain --On Tuesday, February 24, 2004 11:00 am -0700 Thomas M. Schaefer [EMAIL PROTECTED] wrote: Hi all, I have a strange problem. Whenever I plug in the base cord connected to the X100, my DSL service goes down. I DO have a Cisco filter (the one that comes with the product) installed. Has anyone else seen this problem? There was a similar entry in the archives, but it was without a filter. Thanks, Tom Schaefer ___ 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
Re: [Asterisk-Users] Re: DSL (DMT) goes down when X100 plugged in
Looking at the reference design for the chipset used in an X100P a fair chunk of capacitance is slapped straight across the line which would present a significant load to DMT signals. I guess the fax machine introduces some inductance in series with the phone to compensate. I found this link that says a little more about what's in the splitter for the UK marker http://www.adslnation.com/support/filters.php. Iain --On Wednesday, February 25, 2004 12:06 pm -0700 [EMAIL PROTECTED] wrote: My guess was the 100 presented too low an impedence to the line. So, I took an answering machine that had a phone jack on it (pass-through). I plugged the ans. machine into the filter and the 100 into the ans. machine. Everything works now. I can also try a second filter. Thanks, Tom Schaefer ___ 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
Re: [Asterisk-Users] Executing external script
... looks like a case for the System application or AGI. Check out AGI on the VoIP wiki. Iain --On Wednesday, February 18, 2004 12:41 pm +0100 Alessio Focardi [EMAIL PROTECTED] wrote: Hello asterisk-users, just a simple question: I'm looking for a way to execute an external script (php) on the server when an extension is dialed. I have looked around in google without results ... Tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] Voip in the EU
I stuck with this line of argument mainly because the current Ofcom consultation concerns secondary line VoIP. So the customer base is mainly users of Vonage etc. There are, of course many more users of telephony over IP. Iain --On Tuesday, February 17, 2004 8:42 am +0800 Steve Underwood [EMAIL PROTECTED] wrote: Iain Stevenson wrote: The problem with the Ofcom consultation as I see it is that it seems to be regressive wrt to the position now being taken by the FCC. There are probably not many more than 250,000 VoB users worldwide so now is not the time to impose significant market constraints. Why do you quote VoB, when the use of broadband versus other internet connections is totally arbitrary? The figure you quote seems far too low for voice over internet (rather than VoIP, since a lot of the IP is on private nets). I think you will find each of the major producers of VoIPs phone has produced rather more than that. Business users alone, dumping their PBXs, must accounts for millions of lines by now. Some of that traffic goes branch to branch over private nets, but they do a lot of interconnecting with the PSTN too. Regards, Steve ___ 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
Re: [Asterisk-Users] Voip in the EU
The problem with the Ofcom consultation as I see it is that it seems to be regressive wrt to the position now being taken by the FCC. There are probably not many more than 250,000 VoB users worldwide so now is not the time to impose significant market constraints. The new EU regulatory framework actually imposes very few constraints on new service providers in emerging markets such as VoIP being based as it is on the concept of significant market power (SMP). I don't think any carrier has SMP in VoB so the real issue is the extent to which Ofcom tinkers in the interpretation of the rules. Unfortunately they seem to be focusing on the red herrings of emergency service support and lawful intercept - neither of which are of much interest to users. Fixed and mobile services already provide acceptable emergency access. The real issue is the umbrella topic of Universal Service Provision and what the impact of VoIP will be on that. The tone of the Ofcom invitation to the VoB briefing focused on issues that could limit the market rather than promote it. Let's hope that the VoB briefing is followed up by some balanced and broad based consultation. Iain --On Monday, February 16, 2004 5:55 pm + WipeOut [EMAIL PROTECTED] wrote: Linus Surguy wrote: Does anyone know where I can find some more info on the VoIP laws in the EU? VoIP in the EU hasn't been completely sorted centrally (i.e. by the EU parliament), last time they looked at it a few years ago it wasn't perceived to be entranched enough to worry about, I suspect this will change soon. In the UK Oftel put out a guide, which says if you're running VoIP services (i.e. back-end services, so maybe a SIP proxy/registration server or interconnection with the PSTN) you are a Communications Service Provider and covered by the same regulations as a traditional voice provider. Just to clarify this from a different direction, Oftel/Ofcom approach these things by say that they are 'technology neutral', i.e. as standard they don't care how the service is delivered, it is the service that is regulated and not the delivery mechanism. This means in theory the rules for VoIP are the same for copper, wireless, mobile etc. Linus As I understand it that is what the Ofcom VoB discussion next week is all about.. The standard line telco's have to be required to provide a service in an emegency eg during a power failure, but this is impossible for a VoIP provider sine the provider does not have control over the full path or the electricity supply.. That is only one example where VoIP cannot be regulated in the same way as standard telephone services.. In my mind there will have to be separate regulations, there may well be some common clauses but they will still be separate regulations.. Later.. ___ 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
Re: [Asterisk-Users] Voip in the EU
Well, since they restricted attendance to service providers and representatives of consumer organisations I wouldn't be too optimistic for a balanced outcome ;-) Iain --On Monday, February 16, 2004 4:51 pm + WipeOut [EMAIL PROTECTED] wrote: Steve Kennedy wrote: On Sat, Feb 14, 2004 at 02:47:11AM -0500, Ryan Finnesey wrote: Does anyone know where I can find some more info on the VoIP laws in the EU? VoIP in the EU hasn't been completely sorted centrally (i.e. by the EU parliament), last time they looked at it a few years ago it wasn't perceived to be entranched enough to worry about, I suspect this will change soon. In the UK Oftel put out a guide, which says if you're running VoIP services (i.e. back-end services, so maybe a SIP proxy/registration server or interconnection with the PSTN) you are a Communications Service Provider and covered by the same regulations as a traditional voice provider. Steve I am going to an Oftel meeting to discuss VoB regulation next week.. Hopefully this will help to see where it is heading.. Later.. ___ 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
Re: [Asterisk-Users] Digium connectivity issue?
Yes - not much seems to be creeping out of the list servers. Iain --On Friday, February 13, 2004 07:54:50 -0600 Rich Adamson [EMAIL PROTECTED] wrote: Are others seeing hugh delays and/or lack of connectivity to Digium? Rich ___ 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
Re: [Asterisk-Users] Festival: read text from external fil
You can probably use the festival text2wave utility in a cron job to create a speech file from your source text and then use asterisk's Playback function to play it as required. Iain --On Saturday, February 14, 2004 9:41 pm +0100 Lars Fredriksson [EMAIL PROTECTED] wrote: Hello! I wan't to use Festival for reading text from an external textfile - anyone that has a solution for doing that? I can't figure out how I should be able to do that - if it is possible? The textfile contains the temperature and will change every tenth minute - and therefore I can't use include in extensions.conf. Best regards, Lars --- Lars Fredriksson Ockelbo, Sweden mailto:[EMAIL PROTECTED] http://www.fredriksson.net/ ___ 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
Re: [Asterisk-Users] Cisco ATA 186
Search the list - there's a detailed answer on it. I have two of the I1 version (at least that's what they say they are - ProductId: ATA186I1) and they work with UK spec phones. All you need to watch for is that UK phones are three wire and US phones are 2 wire. Maplin sells an adapter to sort this out (Part no. VD36P). Iain --On Wednesday, February 11, 2004 4:54 pm +0100 Dawid Mielnik [EMAIL PROTECTED] wrote: Cisco ATAs come in two types ATA186-I1 with 600 ohm impedance and ATA186-I2 with complex impedance (270 ohm in series with 750 ohm and 150 NF in parallel) What is the difference between the two ? Which one is suitable for Europe ? Thanks, Dave ___ 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
Re: [Asterisk-Users] OS X -- More Specific
--On Monday, February 9, 2004 8:35 am -0700 Erick Schmidt [EMAIL PROTECTED] wrote: When I try to make Asterisk I get the following error: In file included from aescrypt.c:39: aesopt.h:156:22: endian.h: No such file or directory aesopt.h:157:24: byteswap.h: No such file or directory make: *** [aescrypt.o] Error 1 powerbk-g4:/build/asterisk-0.7.2 root# The aes stuff isn't essential for * at the moment so you should be able to comment it out of the Makefile. Some say that zaptel and libpri need to be installed before Asterisk but those packages won't install either. I am very new to Asterisk and do not know the significance of zaptel or libpri. I am trying to install it on a G4 running OS X 10.3.2. You don't need libpri unless you're running ISDN PRI lines into * - so you may not need that library. It's worth getting * going without it first. You will most likely need zaptel so that build needs to be fixed. What sort of errors are you getting? Iain ___ 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] Back to front logging for calls placed through /var/spool/asterisk/outgoing?
That was interesting. Asterisk creates the first cdr entry when the call
file is copied to /var/spool/asterisk/outgoing:
,,271536,callout,,Local/[EMAIL PROTECTED],2,Zap/1-1,Hangup
,,2004-01-25 12:22:54,2004-01-25 12:22:57,2004-01-25
12:22:57,3,0,ANSWERED,DOCUMENTATION
On completion of the call, it generates a second cdr entry:
,,10,home,,Local/[EMAIL PROTECTED],1,SIP/cisco-eca4,Dial,
sip/cisco||tTr,2004-01-25 12:22:57,2004-01-25 12:22:59,2004-01-25
12:23:14,17,15,ANSWERED,DOCUMENTATION
Neither of these entries went into the MySQL backend (via cdr_odbc) so this
has unfortunately only been partially successful.
Iain
--On Saturday, January 24, 2004 12:26:03 -0500 John Todd [EMAIL PROTECTED]
wrote:
Try this: make your outbound call via a Local channel, and see if that
gets logged.
Channel: Local/[EMAIL PROTECTED]
MaxRetries: 0
RetryTime: 60
WaitTime: 60
Context: home
Extension: 10
Priority: 1
and then...
[callout]
exten = _X.,1,Dial(Zap/1/${EXTEN})
exten = _X.,2,Congestion
exten = _X.,102,Busy
exten = h,1,Hangup
JT
Here's an example - placing a call to 271536 from local extension
10. The call file is:
Channel: Zap/1/271536
MaxRetries: 0
RetryTime: 60
WaitTime: 60
Context: home
Extension: 10
Priority: 1
... and the cdr record generated by * on completion of the call is:
,,10,home,,Zap/1-1,SIP/cisco-4edb,Dial,sip/cisco||tTr,
20 04-01-24 16:02:10,2004-01-24 16:02:13,2004-01-24
16:02:26,16,13,ANSWERED,DOCUMENTATION
cisco is the name given in sip.conf for extension 10. I was
expecting a cdr entry for the Zap/1 channel.
Iain
--On Friday, January 23, 2004 21:55:40 -0500 John Todd
[EMAIL PROTECTED] wrote:
Iain -
Brian I believe is correct, and Kannaiyan perhaps is not correct.
Perhaps you can post the actual values in one of your call spool files
so that we can comment on it more clearly. Using the Application:
statement in an outbound spool file will prevent a CDR from being
created; use Context:/Extension:/Priority: methods. If that fails,
then we have a bug.
JT
At 5:59 PM -0600 1/23/04, Brian West wrote:
NO it will log from a spool file if and only if you ref an extension
and not an application.
bkw
On Fri, 23 Jan 2004, Kannaiyan Natesan wrote:
There is no CDR for the call from spool outgoing,
You need to write a patch to solve the same.
Kannaiyan
- Original Message -
From: Iain Stevenson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 8:27 PM
Subject: [Asterisk-Users] Back to front logging for calls placed
through /var/spool/asterisk/outgoing?
I've just noticed that if you start a call by writing a file to
/var/spool/asterisk/outgoing the cdr created on termination logs
the call placed to the local extension - not to the destination in
the PSTN. Hence there is no record of the PSTN number dialled. I
guess most people want
to
log the outgoing portion not the local call leg? Anyone know of a
setting that changes this?
Iain
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Re: [Asterisk-Users] Back to front logging for calls placed through /var/spool/asterisk/outgoing?
Here's an example - placing a call to 271536 from local extension 10. The call file is: Channel: Zap/1/271536 MaxRetries: 0 RetryTime: 60 WaitTime: 60 Context: home Extension: 10 Priority: 1 ... and the cdr record generated by * on completion of the call is: ,,10,home,,Zap/1-1,SIP/cisco-4edb,Dial,sip/cisco||tTr,20 04-01-24 16:02:10,2004-01-24 16:02:13,2004-01-24 16:02:26,16,13,ANSWERED,DOCUMENTATION cisco is the name given in sip.conf for extension 10. I was expecting a cdr entry for the Zap/1 channel. Iain --On Friday, January 23, 2004 21:55:40 -0500 John Todd [EMAIL PROTECTED] wrote: Iain - Brian I believe is correct, and Kannaiyan perhaps is not correct. Perhaps you can post the actual values in one of your call spool files so that we can comment on it more clearly. Using the Application: statement in an outbound spool file will prevent a CDR from being created; use Context:/Extension:/Priority: methods. If that fails, then we have a bug. JT At 5:59 PM -0600 1/23/04, Brian West wrote: NO it will log from a spool file if and only if you ref an extension and not an application. bkw On Fri, 23 Jan 2004, Kannaiyan Natesan wrote: There is no CDR for the call from spool outgoing, You need to write a patch to solve the same. Kannaiyan - Original Message - From: Iain Stevenson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 23, 2004 8:27 PM Subject: [Asterisk-Users] Back to front logging for calls placed through /var/spool/asterisk/outgoing? I've just noticed that if you start a call by writing a file to /var/spool/asterisk/outgoing the cdr created on termination logs the call placed to the local extension - not to the destination in the PSTN. Hence there is no record of the PSTN number dialled. I guess most people want to log the outgoing portion not the local call leg? Anyone know of a setting that changes this? Iain ___ 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
Re: [Asterisk-Users] doublehash patch doesn't work in asterisk 0.7.1
This is similar to the last version and applies against the current cvs. cd asterisk patch -p0 Parking.patch Then the double has transfer should be back. Iain --On Friday, January 16, 2004 6:10 pm -0500 mattf [EMAIL PROTECTED] wrote: Hello, I was using the doublehash.patch that Iain Stevenson had created back in August to change the transfer key from a single hash # to a double-hash #. It always patches properly, but when I went from CVS 2004-01-12 to Asterisk 0.7.1 it doesn't seem to work anymore. I've attached the patch to this email and I use the following command to patch it: patch -p1 ./doublehash.patch Any help would be great. I would like to get this to work with 0.7.1, because we are dependant upon the doublehash patch working. Thanks, MATT--- Parking.patch Description: Binary data
[Asterisk-Users] Back to front logging for calls placed through /var/spool/asterisk/outgoing?
I've just noticed that if you start a call by writing a file to /var/spool/asterisk/outgoing the cdr created on termination logs the call placed to the local extension - not to the destination in the PSTN. Hence there is no record of the PSTN number dialled. I guess most people want to log the outgoing portion not the local call leg? Anyone know of a setting that changes this? Iain ___ 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] cdr_odbc not logging integers eg duration
I tried that - no errors reported. I checked one or two of the SQL calls and none returns an error. I rebuilt and reinstalled mysql and all the ODBC drivers - still no integers written! The direct MySQL driver logs calls fine. So it looks like there's a deeper problem with ODBC to sort out - at least on my system. Iain --On Saturday, January 17, 2004 6:25 pm -0600 Brian West [EMAIL PROTECTED] wrote: Check your tables. I logged everything as integer. set verbose 10 and make a call and watch it.. then do reload and watch the output. It will unload and reload and you can check to make sure your accually connetcing to the database. bkw On Sat, 17 Jan 2004, Iain Stevenson wrote: I've just noticed that since swapping from the direct mysql cdr driver to cdr_odbc, the call duration (and anything else that's an integer) isn't logged - anyone else seen this and know the reason. The cdr_odbc driver gives no error messages and records any string based fields correctly. Iain ___ 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 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] cdr_odbc not logging integers eg duration
I've just noticed that since swapping from the direct mysql cdr driver to cdr_odbc, the call duration (and anything else that's an integer) isn't logged - anyone else seen this and know the reason. The cdr_odbc driver gives no error messages and records any string based fields correctly. Iain ___ 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] People detected as fax machines
A caller to me was this afternoon detected as a fax machine: Jan 15 15:31:17 NOTICE[41997]: File chan_zap.c, Line 3564 (zt_read): Fax detected, but no fax extension ... and then redirected to voicemail. An extract from extensions.conf is attached below. Is there any way to stop * even considering an incoming call on a line as a fax call? Iain bell] include = mailboxes include = day|07:55-23:00 include = night exten = t,1,Voicemail2,100 exten = t,2,Hangup [day] ; set music on hold for parked calls exten = s,1,setmusiconhold,default exten = s,2,responsetimeout,20 ; ring SIP for 20 seconds exten = s,3,Dial,sip/ciscosip/cisco1sip/cisco2sip/cisco3|20|tT ;if nobody answers tell them how to use the voicemail system. ; exten = s,4,Background,vmprompt exten = s,5,Voicemail2,100 exten = s,6,Hangup ___ 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] Major format changes
app_festival currently seems to chop the start of sound it plays back - probably something to do with rtp and maybe the same problem that was present in voicemail prompt plauback. Iain --On Thursday, January 15, 2004 11:16 am -0600 Steven Critchfield [EMAIL PROTECTED] wrote: On Thu, 2004-01-15 at 10:41, Robert Murray wrote: Hi Mark Would it be possible to include a way of streaming audio from memory? For example registering a file type which read from a fifo in memory? I need this for app_theta. (Cepstral TTS) I could copy the code from file.c, but I think it would be better if the same code could be used to avoid duplication. Whats wrong with just creating your frames and handing them off to be dealt with? Maybe you should check out the stuff in app_festival. On Sat, Jun 28, 2003 at 05:48:50PM -0500, Mark Spencer wrote: I've made some major changes to the way Asterisk handles file formats. I'd like feedback from people about any experience they have with these changes. They *may* improve playback performance for people who have had trouble with playback performance in the past. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] Re: newbie ISDN question
Will the driver support big endian systems (PPC) - most ISDN cards seem to ship only with Wintel drivers. I have ISDN but at the moment have to use an analogue interface through a TA. Iain --On Wednesday, January 14, 2004 3:11 pm +0100 Klaus-Peter Junghanns [EMAIL PROTECTED] wrote: Hi, yes, for the home user it's still too expensive. Although it's really cheap if you compare it to other 4 BRI cards on the market. Currently i am polishing the driver for the hfc-s pci a chipset, which i used in numerous el-cheapo ISDN cards (street price around 30 EUR). This will bring zaptel BRI (and even NT mode) to the home user. :) best regards kapejod -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Straße 13 - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Hi, On Wed, 14 Jan 2004 at 12:15, Klaus-Peter Junghanns wrote: The quadBRI card is EUR 600, excluding VAT. this looks like a great piece of hardware, but I think it's too expensive for home users like me who wouldn't really need more than one or two BRI ports. So do you have any plans for a singleBRI or doubleBRI version of this card, or maybe even a variant that comes with a single port preinstalled and three more ports can be added as needed via daughterboards like on the TDM400P? cu Reinhard ___ 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 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 and Festival (* dies with no info)
It may not be you, I think the Festival driver is buggy. Specifically, I've found that the the way in which you pass the text to Festival matters. If I use the Festival () suntax then it won't work. If I use the wrong sort of quotation mark instead of ' there are problems. Asterisk will consume vast amounts of processor resources. However, if I specify the command in a way the Festival app likes then all is OK. Try variants like: exten = 555,4,Festival,'mary had a little lamb' Iain --On Tuesday, January 13, 2004 8:11 am -0500 Doug Raum [EMAIL PROTECTED] wrote: Hello, I have Asterisk running on a RH9 box; Everything seems to be working as it should, except for Festival. Every time that Festival is called from Asterisk, Asterisk silently shuts down. Festival doesn't give any error indication and Asterisk just plain dies without a peep. Festival was installed per the Wiki, using source and patched with festival-1.4.3-diff; it works fine at the console. Asterisk is built from CVS and has been configured per the Wiki as well, including the test extension (555). I start Festival with the festival_server script, then start Asterisk. (snippet from extensions.conf) exten = 555,1,Answer exten = 555,2,Festival(mary had a little lamb) exten = 555,3,Hangup Here's what Asterisk says with -v, calling from SIP 81001 to 555: Asterisk Ready. -- Executing Answer(SIP/81001-e87b, ) in new stack -- Executing Festival(SIP/81001-e87b, mary had a little lamb) in new stack == Parsing '/etc/asterisk/festival.conf': Found == Spawn extension (from-sip, 555, 2) exited non-zero on 'SIP/81001-e87b' ...at this point Asterisk is dead. No segfault, no error message. # cat /var/log/asterisk/messages Jan 7 15:36:49 WARNING[1074416352]: File chan_iax2.c, Line 5466 (set_config): Ignoring port for now # cat /var/log/asterisk/event_log Jan 7 15:36:47 asterisk[5038]: Started Asterisk Event Logger (I capture stderr to asterisk.err) # cat /var/log/asterisk/asterisk.err Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe I'm guessing the ouch comes from mpg123 being surprised that Asterisk is gone. Debug info in syslog seems pretty unhelpful if I use -d: Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 4024 (check_user): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 5098 (handle_request): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 1002 (find_user): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 3417 (build_route): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 304 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 361 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 363 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 379 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 400 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1234379840]: File app_festival.c, Line 410 (festival_exec): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 567 (__sip_ack): Jan 7 15:37:00 asterisk_pbx[5038]: Jan 7 15:37:00 DEBUG[1150495040]: File chan_sip.c, Line 567 (__sip_ack): Jan 7 15:37:01 asterisk_pbx[5038]: Jan 7 15:37:01 DEBUG[1234379840]: File cdr_addon_mysql.c, Line 123 (mysql_log): Jan 7 15:37:01 asterisk_pbx[5038]: Jan 7 15:37:01 DEBUG[1234379840]: File cdr_addon_mysql.c, Line 130 (mysql_log): Jan 7 15:37:01 asterisk_pbx[5038]: Jan 7 15:37:01 DEBUG[1234379840]: File chan_sip.c, Line 1081 (sip_hangup): Festival's info is very minimal, but seems to indicate success: # cat festival_server.log Load server start ./festival_server.scm festival port=1314 wrapper Wed Jan 7 15:36:40 EST 2004 : USING DEFAULT CONFIGURATION wrapper Wed Jan 7 15:36:41 EST 2004 : waiting serverWed Jan 7 15:36:41 2004 : Festival server started on port 1314 client(1) Wed Jan 7 15:37:00 2004 : accepted from localhost client(1) Wed Jan 7 15:37:00 2004 : disconnected ...a process listing after the * crash shows a zombie festival, although Festival will happily take new connections: 5024 ?S 0:00 /bin/sh /usr/local/festival/bin/festival_server 5030 ?S 0:00 festival --server ./festival_server.scm 5065 ?Z 0:00 [festival defunct] I can restart Asterisk again, and do this over and over and over. If I use the -g option to generate a core dump, I never see one generated. Any thoughts on what might be happening here? What am I doing wrong? -- Doug
Re: [Asterisk-Users] SIP and AGI crash...
Looks familiar to me - check this: http://bugs.digium.com/bug_view_page.php?bug_id=695 Iain --On Tuesday, January 13, 2004 4:55 pm + Tristan 'Minty' Colgate [EMAIL PROTECTED] wrote: Hi, I'm trying to use the say-ani agi asterisk-perl script and am experiencing crashes, I am also experienceing problems with the test-agi scripts shipped with asterisk. The clearest demonstration of the problem is that if I dial extension 125 configured as... exten = 125,1,Ringing exten = 125,2,Wait(3) exten = 125,3,Answer exten = 125,4,Wait(2) exten = 125,5,AGI(agi-sayani.agi) exten = 125,6,Hangup I can crash the asterisk server by hanging up during the call, if I leave the call to complete and let * hang up then everything seems fine. Asterisk does not crash if I am running from the console, only if asterisk has been started in the background (it does still crash if I am attached via asterisk -r at the time the call is hung up). Using the agi test script (on extension 126, same config as above) I get the following... *CLI -- Executing Ringing(SIP/-08135e80, ) in new stack -- Executing Wait(SIP/-08135e80, 3) in new stack -- Executing Answer(SIP/-08135e80, ) in new stack -- Executing Wait(SIP/-08135e80, 2) in new stack -- Executing AGI(SIP/-08135e80, agi-test2.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-test2.agi AGI Environment Dump: -- accountcode = -- callerid = SNOM Phone 1543 8552 -- channel = SIP/-08135e80 -- context = sip-gw -- dnid = unknown -- enhanced = 0.0 -- extension = 126 -- language = en -- priority = 5 -- rdnis = unknown -- request = agi-test2.agi -- type = SIP -- uniqueid = 1074011198.0 1. Testing 'sendfile'...PASS (0) 2. Testing 'sendtext'...PASS (0) 3. Testing 'sendimage'...PASS (0) -- Playing 'digits/1' (language 'en') -- Playing 'digits/hundred' (language 'en') -- Playing 'digits/90' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/million' (language 'en') -- Playing 'digits/8' (language 'en') -- Playing 'digits/hundred' (language 'en') -- Playing 'digits/30' (language 'en') -- Playing 'digits/7' (language 'en') -- Playing 'digits/thousand' (language 'en') Jan 13 16:26:50 WARNING[1116941120]: chan_sip.c:471 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) == Spawn extension (sip-gw, 126, 5) exited non-zero on 'SIP/-08135e80' -- Executing Hangup(SIP/-08135e80, ) in new stack == Spawn extension (sip-gw, h, 1) exited non-zero on 'SIP/-08135e80' PASS (-1) 5. Testing 'waitdtmf'...FAIL (unexpected result '') 6. Testing 'record'...FAIL (unexpected result '') 6a. Testing 'record' playback...FAIL (unexpected result '') == Complete == 7 tests completed, 4 passed, 3 failed == The test seems to stop half way through. I am not entirely sure that these two issues are actually related though as I don't see any of the warning from chan_sip if I hang up during a call to the say-ani script. I don't seem to be getting a core dump, are there any known issues with AGI at the moment? Voicemail, SayUnixTime and everything else is working fine. -- Tristan 'Minty' Colgate [EMAIL PROTECTED] | ICQ #154577755 --- I don't mean to sound bitter, cold, or cruel, but I am, so that's how it comes out - Bill Hicks ___ 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
Re: [Asterisk-Users] Cisco Gear
Prices? Are we talking a 7960 for the price of a SNOM? Iain --On Friday, January 9, 2004 6:00 pm + Adthrawn [EMAIL PROTECTED] wrote: Hi, I know it's not really the place, but if anybody in the UK (or US) is interested, I'm clearing out lots of new Cisco stock... 7970G's (colour LCD), 7960G's, 7940G's, 7920G's (wireless IP phone), 7935's (conference phone) and 3550-24-PWR switches. I also have boxes of 7914's, the single-7914 foot stand and double-7914 foot stand (these are required to connect a 7914 to a 7960G). And some useful locking and non-locking wallmount brackets for 79xx range. We also have lots of PSU's for the whole 79xx range. I'll now feel ashamed, and sink into my seat :-) Best, Ad. ___ 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
Re: [Asterisk-Users] Mailing list growth
--On Thursday, January 8, 2004 11:25 am +0100 Olle E. Johansson [EMAIL PROTECTED] wrote: Well, mailing list growth is not only a good thing. It's getting almost impossible to handle. As I've stated before, we need to change Asterisk.org so we can help people in a better way and avoid a lot of the repeating questions on the mailing list. There's a lot of people unsubscribing, just because of the amount of messages. Asterisk.org needs an FAQ, more documentation on line and ... I think people tend to migrate to the list that seems the most active and post any question to it - on the basis that it has the greatest number of eyeballs hovering over it hence increasing the probability of a rapid response. Not much you can do about that. Asterisk, or more correctly, Digium has hardware as well as software components. So the list has a mix of hardware support issues as well as software support issues - I think that adds a lot to the volume. For me, the wiki has proved to be invaluable as I've experimented with AGI and the manager interface (I must add some comments to it on this .). However, it really does need to be complemented by better documentation so that it's easier to understand exactly how some of the advanced features are used. My usual error is to misunderstand how some feature of * responds because the documentation is ambiguous ... then I plunge into the code and waste a lot of time figuring out what is actually happening. From wet and windy Southern England, Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A Note to GS users..
--On Wednesday, January 7, 2004 5:24 pm + WipeOut [EMAIL PROTECTED] wrote: The GS phones have a setting for Voice Frames per TX with a default value of 10.. This causes the phone to use a 100ms packet size and Asterisk is set to use a 20ms pachet size.. The result is a choppy sound when calling out over the PSTN (specifically in my case ove an X100P).. The solution is to set this value to 2 which will clear up the choppy sound.. This seems to be the default value for the latest firmware - so no need to check recently purchased phones. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to monitor calls initiated by .call file using manager interface?
You can use the asterisk management interface to query for extension status etc - see http://www.voip-info.org/wiki-Asterisk+manager+API. You may need to supply a channel number for the device you want to monitor. This is usually derived from the name you supplied for the extension in the relevant .conf (eg sip.conf). Iain --On Monday, January 5, 2004 11:05 pm -0500 Serge Mankovski [EMAIL PROTECTED] wrote: Hi Here is my problem: I initiate a conference call by placing several .call files into /var/spool/asterisk/outgoing/ directory Asterisk starts calls and I can see events in the manager interface. At the same times there are other calls going on and there are many more events in the manager interface. How can I identify events that are related to the calls started via spool? I tried to pass additional variables in the call using SetVar: statement, but they do not get propagated into events. Is there a way to do what I need? Thank you Serge _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=dept/featurespgmarket=en-caRU=http%3a%2f%2fjo in.msn.com%2f%3fpage%3dmisc%2fspecialoffers%26pgmarket%3den-ca ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail Out call
It is a problem - but the call recording is saved by * when you hang up.
So you need to look for new files in whichever directory the call
recordings are saved and pick them up eg with a script.
Iain
--On Sunday, January 04, 2004 12:07:35 -0500 Kevin [EMAIL PROTECTED]
wrote:
There was a post in the 'wiki' for an application to provide an outcall
when there is a voicemail is left on asterisk. I am having a problem
that this application will only work if the caller presses the pound sign
at the end of recording. As most people just hang up, this application
isn't working. Can any offer suggestions to accomplish this out call?
http://voip-info.org/wiki-Asterisk+tips+callback
[macro-leave_voicemail]
; Leave a voicemail message, then do post-processing.
; o Call configured phones, with an announcement that a message
; is waiting, and the option to listen to the voicemail(s)
;${ARG1} = u or b for 'unavailable' or 'busy' message
;${ARG2} = mailbox
; ${ARG3} = Call user flag
; USAGE:
; exten = s,15,Macro(leave_voicemail,u,310,1)
exten = s,1,ResponseTimeout(30)
exten = s,2,Voicemail2(${ARG1}${ARG2})
exten = s,3,GoToIf($[${ARG3} = 0]?s|5)
exten = s,4,system(${SCRIPTS_DIR}/voicemail_callback.sh ${ARG2})
exten = s,5,NoOp
exten =
h,1,GoToIf($ANBSP;CLASS='WIKI'NBSP;NBSP;HREF='${ARG3}NBSP;=NBSP;0'
${ARG3} = 0?h|3) exten =
h,2,system(${SCRIPTS_DIR}/voicemail_callback.sh ${ARG2}) exten =
h,3,NoOp
exten =
t,1,GoToIf($ANBSP;CLASS='WIKI'NBSP;NBSP;HREF='${ARG3}NBSP;=NBSP;0'
${ARG3} = 0?t|3) exten =
t,2,system(${SCRIPTS_DIR}/voicemail_callback.sh ${ARG2}) exten =
t,3,NoOp
Thanks,
Kevin
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Re: [Asterisk-Users] Re: Grandstream Quality Survey.... :P
--On Monday, December 29, 2003 11:28 am +0100 Cees de Groot [EMAIL PROTECTED] wrote: Lubomir Christov [EMAIL PROTECTED] said: Yes, I know that the Grandstream firmware have problems (I have here 15 phones with some beta version already installed :( and waiting for bug fixing in the new beta) but the stable version 1.0.3.81 is working just perfect. Here too. Would be interested to learn what the problems are with 1.0.3.81. And if people complain about beta firmware, well... I guess that's why they call it beta, not? .. except that Grandstream are shipping new phones with the beta code ;-) Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Grandstream Quality Survey.... :P
--On Monday, December 29, 2003 11:58 am -0700 [EMAIL PROTECTED] wrote: Lubomir Christov [EMAIL PROTECTED] said: Yes, I know that the Grandstream firmware have problems (I have here 15 phones with some beta version already installed :( and waiting for bug fixing in the new beta) but the stable version 1.0.3.81 is working just perfect. Here too. Would be interested to learn what the problems are with 1.0.3.81. And if people complain about beta firmware, well... I guess that's why they call it beta, not? .. except that Grandstream are shipping new phones with the beta code ;-) Iain I just got 2 101s with 1.0.4.17 pre-installed which means I can't go back to 1.0.3.x. I really haven't had too many problems with it yet but I haven't used them much which I guess makes it bad. Shrug. Nothing major. I haven't had problems either wit basic phone calls. The issues regarding DTMF handling and the Early Dial feature noted on this list are present in 1.0.4.17 though cos that's the version of firmware I have and I've tested these issues. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] outcall notification
Maybe you just need to dump a file to the spool directory that has your phone number and an asterisk extension that goes to a voicemail check. You'd still need to patch app_voicemail to create the call file. Iain --On Sunday, December 28, 2003 4:07 pm -0500 Kevin [EMAIL PROTECTED] wrote: Has anyone implemented an outcall notification when there is a voice message waiting? I would like to have the system notify me of awaiting voice messages by a telephone call rather than an email notification. I would imagine that a call could be dumped into the asterisk spool directory, but I'm not sure how I would monitor for messages waiting. Has anyone implemented such a feature for asterisk? I did a google and wiki search with no information available. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DevKitLite compiles but won't load modules or run asterisk
--On Thursday, December 25, 2003 9:13 pm -1000 Ron Fox [EMAIL PROTECTED] wrote: Also, is there a script or makefile target that will fully un-install asterisk, zaptel, zapata and libpri so that I can try again? You could install the utility checkinstall. It creates a RPM for software that installs via make install - as asterisk does. You can then remove the software using rpm -e Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD problems
--On Wednesday, December 24, 2003 07:12:05 -0600 denon [EMAIL PROTECTED] wrote: I've got it running through Asterisk - all working fine from a SIP standpoint. I can dial FWD numbers like 612/613/etc and everything works. However, if I dial *18005551212 or *408xxx (say, a USA number), I either get a fast busy or a This service is only available to FreeWorld Dialup members. I have exactly this problem and posted a bug report to FWD about a week ago - no response yet. It's bizarre that FWD recognises you to dial another user but not to call outside their network. Sounds more like a FWD problem than a * problem to me. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FWD problems
--On Wednesday, December 24, 2003 6:35 pm +0100 Arnold Ligtvoet [EMAIL PROTECTED] wrote: Read the fwd announcement. Jeff Pulver mentioned the fact that * users cannot use the free holiday calls, since FWD cannot be sure that * is not being used by more than 1 user at the same time. Where in this announcement: On Free World Dialup, go ahead and dial: *1 (area code) Number. We have arranged to pick up the costs to allow members of the Free World Dialup community to place calls into the US and Canada for Free during the 2003 holiday season. While the offical press release will follow later today or tomorrow, you can help out in the beta-trials of this holiday gift today. Feel free to share the holiday spirit and cheer. :-) .. does it say * cannot be used? Remember, I tried this a week ago and got the this service is available to FWD members only message. Pulver posted the message mentioning the restriction on 21 December - I've been waiting since December 18 for a reply to my original report of a problem. Still, there seems to be a you get what you pay for theme to many of today's posts and this clearly applies to support on FWD. Naybe we should remove the signature from * that enables FWD to identify * systems :-) Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FWD problems
--On Wednesday, December 24, 2003 10:06 pm +0100 rnc Info Lists [EMAIL PROTECTED] wrote: Still, there seems to be a you get what you pay for theme to many of today's posts and this clearly applies to support on FWD. Naybe we should remove the signature from * that enables FWD to identify * systems :-) That certainly seems the case for today's theme... It is certainly the right of any company or person to define the rules of their service. Since I don't pay for either Asterisk or FWD then I appreciate the service that is provided and try not to crusify them when things don't go right. This entire VoIP is still rather experimental. If I want guaranteed service then I'll pay some provider for it... THEN.. and only then will a service level be expected. That's fair comment but I think FWD should have put a correct message on their system for asterisk users. It wouldn't have taken much effort. FWD and * complement each other and should benefit from each other's success. Indeed * is cited on the FWD web site and mentioned by Jeff Pulver at his VON events. It seems a little unfortunate that FWD is assuming all * systems are a front for hundreds of users and banning them. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Festival sounds like a steam engine
I tried running the festival app today with little success. I have a working festival installation that does TTS to the linux sound output perfectly. With * it just produces a sort of hissing sound. The length of hissing is proportional to the length of text string that it is given to speak. Since I'm running on a PPC system I fear the dreaded endian problem is to blame and that app_festival may need changing. Has anyone else experienced similar problems? Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BT launches consumer VoIP product ...
VoIP watchers may like to take a look at this: http://www.btbroadbandvoice.com/broadband_voice/bb_voice_home.html BT has launched a consumer VoIP service in the UK using ATA 186s (judging by the picture). Now if only I could connect the service to my * server without the ATA Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] BT launches consumer VoIP product ...
Not quite - I want to use SIP directly from * - I don't need a locked ATA186 as a paperweight ;-) That is, assuming BT locks the config as Vonage does. Iain --On Tuesday, December 9, 2003 3:59 pm + Senad Jordanovic [EMAIL PROTECTED] wrote: You can!!! :) Use one of those FXS to FXO converters found at eBay, connect X100P to the FXO port and you all setup. Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vonage sending Motorola gear now?
--On Sunday, December 07, 2003 09:36:14 -0500 TeleSIP [EMAIL PROTECTED] wrote: Its the VT1000 http://broadband.motorola.com/catalog/productdetail.asp?ProductID=212 We have looked everywhere for it but looks like no distributor sells it right now. Maybe because it's a new variant of the VT1000. PacketCable doesn't use SIP (it uses a derivative of MGCP) so the product may not yet be shipping with the SIP code. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Request for debug message in ENUM code
I've been tinkering with ENUM and found that the lack of a debug message in enum.c that says it has actually succeeded in resolving an address is a bit of a nuisance. It makes it difficult to see if failures with ENUM are due to problems with parsing NAPTR records (in enum.c) or mistakes in extensions.conf An extra line of debug information would be much appreciated! Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Feedback with X100P and SIP fwd.pulver
Well, SIP to SIP with no intervening analogue should produce no echo at all. Echo on SIP to analogue calls has been covered extensively on this list. Do a search on echo. Iain Hello: I have installed *. I configured my SIP account and my X100P. But when I call from SIP or from PSTN. The SIP extension hear an echo voice of its conversation. Anyone can help me??? Thanks, voipfan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wireless
AFAIK the 7920 needs CallManager to work - if you haven't got that you'll have to wait for Cisco to make a general purpose version - or maybe buy a Pulver phone http://www.pulverinnovations.com/ - assuming that works with * Iain --On Monday, November 17, 2003 6:31 am -0500 Jeremy McNamara [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: Has anyone got a mobile wireless phone working with * yet Is it possible to use the Cisco 7920 with skinny Not sure, send me one and I'll test it for you. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bad echo on outgoing calls
Yes - the aggressive suppressor does tend to clip speech although I don't think it is half duplex. The MEC3 echo suppressor seemed to be heading in the right direction but last time I tried it it went funny after a while causing speech interruption. Iain --On Saturday, November 15, 2003 16:23:00 -0800 Ed Rubright [EMAIL PROTECTED] wrote: There was a comment made last week in this list that with echo cancellation set as MARK2 and aggressive suppressor enabled the line would no longer be full duplex! Has anyone actually noticed this? If so, does it actually cause a problem during a normal conversation? Thanks, Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Gillham Sent: Saturday, November 15, 2003 1:59 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Bad echo on outgoing calls Andrew Joakimsen wrote: The X100P cards have horrible echo problems. I've heard talk about this being fixed, but havent seen anything done about it. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Larry D. Black Sent: Saturday, November 15, 2003 3:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Bad echo on outgoing calls I have just installed and configured asterisk I have been playing with software phones and an analog phone plugged into a TDM card. I have one line coming in on a X100P card. My X100P works quite well if I don't adjust the gain. Unfortunately it is a bit on the quiet side without the adjustment. I'll test it out with the echotraining and the gain settings. In the past with gain enabled, the echo would correct after 5-10 seconds of conversation. This is with MEC2, and I tested with and without the aggressive suppressor. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bad echo on outgoing calls
You'll probably need clean builds of zaptel and asterisk - I tried with updates earlier today and the echotraining option wasn't recognised until I did a complete clean install. Iain --On Saturday, November 15, 2003 13:59:13 -0800 Andrew Gillham [EMAIL PROTECTED] wrote: Andrew Joakimsen wrote: The X100P cards have horrible echo problems. I've heard talk about this being fixed, but havent seen anything done about it. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Larry D. Black Sent: Saturday, November 15, 2003 3:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Bad echo on outgoing calls I have just installed and configured asterisk I have been playing with software phones and an analog phone plugged into a TDM card. I have one line coming in on a X100P card. My X100P works quite well if I don't adjust the gain. Unfortunately it is a bit on the quiet side without the adjustment. I'll test it out with the echotraining and the gain settings. In the past with gain enabled, the echo would correct after 5-10 seconds of conversation. This is with MEC2, and I tested with and without the aggressive suppressor. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Apple implementation
I have been running asterisk on an old PowerMac 9600 and YellowDog Linux for about a year now. Asterisk software builds fine most of the time - there seem to be some trivial issues with the Makefiles for codecs at the moment. I have an X100P card as the PSTN interface. I suspect that the interface cards are likely to be your biggest problem - drivers supporting big endian systems are needed. I don't know whether all the Digium drivers do. ISDN cards from AVM and Eicon are not suitable for PPC Linux. Iain --On Wednesday, November 5, 2003 9:17 am -0700 Charles Hatchette [EMAIL PROTECTED] wrote: I am new to Asterisk and Digium card implementation issues. My VAR is strongly recommending using Apple hardware and Yellow Dog Linux for my telephony project, because of his familiarity with this OS. Is the PowerPC an appropriate and stable hardware platform for Digium/Asterisk development? Charles Hatchette [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to escape #
I'll own up to a patch - bug report 110. However, Mark peremptorily
dismissed my suggestion putting forward a solution I find illogical. I
guess more people need to ask for this feature!
I think my original patch was a bit over-engineered. The one below is
simpler.
Iain
--- res_parking.c.orig Sun Aug 24 16:57:10 2003
+++ res_parking.c Sat Sep 27 10:43:17 2003
@@ -25,6 +25,7 @@
#include asterisk/musiconhold.h
#include asterisk/config.h
#include asterisk/cli.h
+#include asterisk/indications.h
#include stdlib.h
#include errno.h
#include unistd.h
@@ -214,6 +215,7 @@
struct ast_channel *transferer;
struct ast_channel *transferee;
char *transferer_real_context;
+int ms;
/* Answer if need be */
if (ast_answer(chan))
@@ -274,6 +276,13 @@
transferer = chan;
transferee = peer;
}
+//IAS
+ ms = 250; /* ms */
+ms = ast_waitfordigit(transferer, ms);
+ if( ms != '#')
+
ast_playtones_start(transferee,0,!941+1477/50,!0/50,0);
+ else {
+
/* Use the non-macro context to transfer
the call */
if(strlen(transferer-macrocontext))
@@ -381,6 +390,7 @@
if (option_verbose 1)
ast_verbose(VERBOSE_PREFIX_2 Hungup during autoservice stop on '%s'\n,
transferee-name);
}
+ }
} else {
if (f (f-frametype == AST_FRAME_DTMF)) {
if (who == peer)
--On Monday, October 20, 2003 8:52 am -0700 John Todd [EMAIL PROTECTED]
wrote:
At 3:42 PM +0200 10/20/03, Louis-David Mitterrand wrote:
Hi,
This morning I found myself stumped when a remote interactive system
asked me to enter some identification followed by the # key, and my
local Asterisk interrupted with Transfer?.
Is there a way to escape the pound key, short of disabling transfers?
Cheers,
--
Make it idiot proof, and somebody will make a better idiot.
There is a patch for this available, I seem to recall. Look through the
archives, and search for ## - someone made it so that the transfer
feature would only work after hitting # twice. A very cursory search of
the bugtracker didn't find that patch - can someone search more
diligently, and then submit it if they find the original code?
JT
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Re: [Asterisk-Users] Cisco 7920 phone
--On Monday, August 18, 2003 10:31 pm +1200 Roger De Salis [EMAIL PROTECTED] wrote: Interesting menu options implying mechanisms to take the 11 channels of WiFI, and dedicate 1-3 for voice, and turn the rest over to data. There were some rumours that they only work on Cisco Aironet base stations They work fine on DLink, Kamaguza, and Uncle Tom Cobblies base stations... I suppose you need the Cisco Aironet for QoS support on the WLAN. Performance may not be good on a highly loaded base station - do you have any test results? The other problem with the 7920 is that you currently need CallManager to host it. It would be bad news if Cisco implemented SIP but kept the requirement for CallManager to host the phone. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with Wildcard 100XP and hangup signal
It should work with the standard PSTN but you can get problems if you connect a X100P to a PBX or ISDN TA. Try editing the wcfxo.c file and enabling support for ZERO_BATT_RING (uncomment the #define) then rebuild and reinstall the zaptel modules - you will need to unload and reload the wcfxo and zaptel modules. Iain --On Tuesday, August 12, 2003 12:57 pm +0200 Emmanuel Bergmans [EMAIL PROTECTED] wrote: Hi, We are currently testing Asterisk with Wildcard 100XP and serveral Cisco ATA Box. Everything works great except that the card does not detect the hangup signal. We are using a standard Belgian PSTN line. I have not found anything about a be zone (only us, fr, de, nl, ...). Does someone experience the same problem? Do I need to create a new zone be (and how to do that)? Another small question. Does anybody have any experience with T400P and E400P on the Belgian phone network? Thanks in advance for your help. Perceval Helpdesk Team [EMAIL PROTECTED] - --- Perceval Technologies sa/nv Rue Tenbosch, 9 B-1000 Brussels BELGIUM Tel: +32-2-6409194 Fax: +32-2-6403154 URL: http://www.perceval.be/ E-mail for general information: [EMAIL PROTECTED] E-mail for technical information: [EMAIL PROTECTED] - --- This e-mail message contains legally PRIVILEGED and CONFIDENTIAL information intended for the use of the addressee only. If you are not the intended recipient of this message, please notify the undersigned by telephone or e-mail reply and destroy this message and any attachments. Any views or opinions presented in this e-mail are solely those of the author and do not necessarily represent those of Perceval. The integrity and security of this message cannot be guaranteed and it may be subject to data corruption, interception, unauthorised amendment, viruses and unforeseen delays, for which we accept no liability. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FXO mode
Assuming this is on incoming calls, the most usual source of the problem is that the telco exchange either doesn't send a disconnect pulse or the wcfxo driver doesn't recognise the format used. I've unfortunately forgotten the exact situation but, when a call finishes, a telco exchange in the US will momentarily remove/reverse line power. That tells the fxo card, hence *, that the call is over. If such a pulse isn't present * assumes the call is still in progress. This has been discussed on the list in the past so it's worth googling. Iain --On Wednesday, August 13, 2003 12:09 pm -0400 Joe Antkowiak [EMAIL PROTECTED] wrote: I've had this happen with the x100p and analog phones as well... When I moved to a t1 and a channel bank, the problem never happened again... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Meyers Sent: Wednesday, August 13, 2003 12:04 PM To: Asterisk List Subject: Re: [Asterisk-Users] FXO mode On Wed, 2003-08-13 at 09:46, Dave Cotton wrote: I've had a few problems with my system holding the line after a call has been made, just now I rebooted and noticed the following in /var/log/messages When you say holding the line, do you mean that asterisk still believes a channel is in use even after you hang up? If so, I've seen the same thing happen several times with the X100P. If I do show channels it will show one of my SIP phones connected to one of the outside lines, but if I check that SIP phone, it is not in use, and there is no way to re-activate the channel from the SIP phone. Running soft hangup zap channel will hangup the channel (you don't need to reboot). I'm not entirely sure what causes it. So far, I've only seen it happen from 2 of our 9 SIP phones, but they're the ones most often on the phone. It always involves an outside line, so I believe the X100P is the problem, but I can't be sure. What other information can I gather to pinpoint the problem? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Leftover Budgettone issues
I was in a call through an ATA 186, * and the PSTN today when someone dialled me over FWD. I got a tone in the earpiece more than once which was jolly annoying. Is this the problem you're getting? I think an option to turn this tone off is needed. Iain --On Thursday, August 7, 2003 7:51 am -0600 Steve Meyers [EMAIL PROTECTED] wrote: On Thu, 2003-08-07 at 01:56, Brian Capouch wrote: 2. This phone does not act like all my others do when I am talking and a call comes in. Instead of the jarring ADSI !!!BOING!!! followed by a series of call waiting beeps, instead I get a ringing tone in the earpiece which is audible to the other party as well. If you find out, please let me know! I've tried all sorts of settings to make it stop that. I'd like to just make it not support call waiting at all on the SIP connection, that would be easiest, but I can't find a way to do it. The BudgeTone configuration doesn't seem to be able to turn this off, either. Hopefully they'll fix this soon... Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
--On Thursday, August 14, 2003 12:58 pm +0200 Dave Cotton [EMAIL PROTECTED] wrote: Last night I posted showing that the problem is repeatable and only occurs in one certain circumstance. I think it is within voicemail.c. If the caller exits voicemail by pressing # the line is dropped correctly, if they just hang up voicemail continues to record. I put some debugging statements into voicemail.c and I think that a condition statement is never reached so the line is held up. I'm just a little concerned that this is patching the symptoms without identifying the correct cause. If a caller presses '#' that tells asterisk the call is over and can be destroyed. This is a very different process from the call being terminated by a signal sent from the remote exchange. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does Wildcard x100p support BT Caller ID inUK?
The chipset used in the X100P - at least the one I have - is designed for the US/Japan market only. The reference design in the datasheet for the chipset does not include facilities for the detection of line voltage reversal. Hence the only way to detect caller ID sent before ringing would be to constantly scan for it - wasteful in processor resources. There is another chipset in the same family that supports full international operation. The datasheet for this describes a procedure for detecting line voltage reversals and capturing caller ID. I guess we need Digium to look into an international version of the X100P based on this chipset. Iain --On Saturday, August 09, 2003 19:39:07 +0200 Andy Powell [EMAIL PROTECTED] wrote: Mark, if the capability for line reversal detection is in the hardware (X100P) then does this mean that the detection of DTMF style caller-id as used in the following countries would ber trivial? or am I hoping too much... Finland, Denmark, Iceland, the Netherlands,India, Belgium, Sweden, Brazil, Saudi Arabia, Uruguay and Japan Imagine all those happy people!! Andy On 09/08/2003 at 10:44 Mark Spencer wrote: Maybe you could open a bug for it, and attach the specs / a link to those specs? Also, I suggest you reply to this message: That's a great idea. The other thing is that we have to detect polarity reversal or we'll constantly be scanning for CID. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
