Re: [Asterisk-Users] list proposal
I think that's a very good idea. When I started to become active in * last December the list was much less congested and Mark usually responded to requests, comments and patches within a few hours. Now things are clearly taking off - good for * and Digium but it's sort of losing the community spirit. Splitting the lists by the channel drivers seems to be a good idea but I think there needs to be a strong link with the development team. In the case of SIP, it is clearly becoming the protocol for VoIP and many people would like the channel driver enhanced or may have patches for it. I'm not sure the bug tracker is the right place for this. It would be better to have a SIP list moderated by a developer where changes could be discussed. Doubtless the same reasoning could apply to ISDN, IAX etc. Iain --On Friday, August 08, 2003 13:25:10 -0500 Steven Critchfield [EMAIL PROTECTED] wrote: With the increased traffic as of late, I'm wondering if it is time to split the list again. Specifically I am wondering if it should be split along the various VoIP protocols and zap hardware, then leave a general list that does configuration other than VoIP related? The hope is that those asking SIP or H323 questions could get help from the various supporters while the main list can deal with transport neutral content like extension logic and voicemail configs. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] ztdummy usb-ohci?
The sort answer is no. The ztdummy code is written specifically for usb-uhci and usb-ohci operates in an entirely different way. However, there is an alternative to ztdummy that uses the real-time clock. Take a look at zaprtc from here http://www.junghanns.net/asterisk/page1.html Iain --On Friday, August 01, 2003 14:52:43 -0700 [EMAIL PROTECTED] wrote: Is it possible to get ztdummy working with usb-ohci? Thanks, Justin ___ 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] Patch - transfer with two rather than one #
Here's a patch that changes the behaviour of # transfers in asterisk. A single # is transferred to the remote phone/system. Two # in quick succession will trigger a transfer. This is very useful for users who have basic analogue phones connected to an ATA 186. For example, when calling a remote conference or IVR system you often want a single # to be sent to the remote system - not to trigger a transfer. I'd like to enhance this so that the double hash transfer can be enabled on a per phone basis - but I think that needs the sip channel structures pulling out into header file. This patch has had only minimal testing so use at your own risk! Iain doublehash.patch Description: Binary data
RE: [Asterisk-Users] RTP codec 13 received - Ciscoincompatibilit y?
.. poking head above parapet, venturing correction .. RTP payload type 13 is comfort noise viz http://www.iana.org/assignments/rtp-parameters whereas payload type 19 is reserved. Maybe Cisco is right ;-) I believe * has a partial implementation of comfort noise but that it's not complete yet. I found I could ignore the error messages with my Cisco ATA 186s. Iain --On Thursday, July 31, 2003 9:46 am +0100 Skuse, Phil [EMAIL PROTECTED] wrote: I have a similar setup to you and get the same message regularly. I don't think it's the cause of your problem. I did some research on it a while ago: IIRC the cisco uses codec 13 for silence suppression whereas asterisk (correctly) uses codec 19. The router can be configured to use 19 also, but I didn't bother. I'm sure somebody will correct me if I'm wrong about this. My system does not have the problem you describe. I can call from a SIP softphone, through asterisk , through the cisco and out to our meridian system or the PSTN. In fact, it works very well. Are you sure that you have the dial-peers on the router configured correctly? -Original Message- From: Cerrajetto [mailto:[EMAIL PROTECTED] Sent: 31 July 2003 09:09 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RTP codec 13 received - Cisco incompatibility? Hello, In our SIP network, Asterisk is the central PBX, and it routes calls to the PSTN thru a Cisco Router - IOS 12.2(11)T9. If a client softphone calls directly via Cisco to the PSTN, the call works successfully. If the client softphone calls via Asterisk to other SIP internal extension, it work fine too. The problem is when a client calls an Asterisk extension, and Asterisk transfers the call (via SIP) to the Cisco: - Pingtel (192.168.1.10) calls [EMAIL PROTECTED] (Extension 300 in Asterisk) - Asterisk transfers to [EMAIL PROTECTED] (Cisco GW) - Cisco tries to call to PSTN (666554433) In that context, Asterisk generates this message while ringing: NOTICE[540685]: File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 13 received The PSTN recipient's phone rings. The client does not receive the typical intermittent tone/signal that means the recipient's phone is ringing. When the recipient answers, the call is inmediantly finished. Maybe a short Hello can be listened. Asterisk shows a response back from Cisco: Bad Request - 'Invalid IP Address' In sip.conf, Asterisk is forced to use g711ulaw. I've tried other codecs with no success. What is the real problem?. Is it a RTP problem with codec 13, o a SIP problem?. Is there a Cisco-Asterisk incompatibility?. This is the sequence generated by Asterisk: -- Registered SIP 'pingtel01' at 192.168.1.10 port 5061 expires 500 -- Executing Dial(SIP/pingtel01-af0d, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] NOTICE[573453]: File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 13 received NOTICE[573453]: File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 13 received -- SIP/192.168.200.200-a3d2 answered SIP/pingtel01-af0d -- Attempting native bridge of SIP/pingtel01-af0d and SIP/192.168.200.200- a3d2 -- Got SIP response 400 Bad Request - 'Invalid IP Address' back from 192.168.200.99 == Spawn extension (default, 003, 1) exited non-zero on 'SIP/peter-af0d' Thank you very much, Mark Cerrajetto. ___ 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] moh/playback for non-zap interfaces
I think the quality for music playback on my SIP stuff is pretty good. The real sound problem is in the voicemail access. I very often get sound dropouts when * is reporting the number of new or old messages. Iain --On Saturday, July 26, 2003 10:39 pm -0500 Mark Spencer [EMAIL PROTECTED] wrote: I've merged some changes from Michael Manousos that should improve sound quality on non-zap channels, including music on hold. I'd like to hear back on or off list about your experiences with the new code. Thanks! 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
Re: [Asterisk-Users] Call transfer on ATA186
The basic call transfer functions, set with the T and t options to the dial application and triggered by pressing a # work fine for me. Make sure that you have set the DialPlan on the ATA 186 so as not to grab the # (ie look for any # character pairs and change the second character or remove it). Iain --On Monday, July 28, 2003 6:58 pm +0300 Dan [EMAIL PROTECTED] wrote: Hi, The call transfer function in Asterisk seems to work in a way which does not permit to ATA186 (or any othet hardware phone with only pne line) to have this feature. If someone tries to transfer a call to an ATA186 based extension, the call is transferred to the correspondent voice mailbox, because the first extension tries to call the last one after the transfer is initiated and during this time ATA is bussy. It is something that can be done to solve this issue? Someone else using Call Transfer on ATA186? They are some special switches available for Dial command in order to get Call Transfer workig on ATA? Thanks, Dan ___ 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] ISDN Fritz RedHat 8.0
Assuming it is a suitable Fritz card your best bet is to get the CAPI library/driver from AVM and then check this out http://www.junghanns.net/asterisk/ - chan_capi is reportedly the best performing ISDN channel driver for asterisk, although I personally haven't used it ;-) Iain --On Sunday, July 27, 2003 8:25 pm +0100 Stuart Hirst [EMAIL PROTECTED] wrote: Has anyone got the BT Speedway (AVM Fritz) card working on a RedHat 8.0 system with *. If so could someone give me some pointers on getting the right sequence of installing the drivers and which versions to use. Thanks, Stuart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD no longer works.. but nothing has changed? Wierd DEBUG errors.
There was a thread on FWD failures yesterday and indeed it didn't work for me at 9:00 in the morning but by 10:30 all was fine - I'd made no changes to *. It looks as though there's some tinkering going on at the FWD end. Iain --On Thursday, July 24, 2003 12:32:00 -0400 Leif Madsen [EMAIL PROTECTED] wrote: I'm wondering if anyone else has gotten something similer to this? I had FWD working fine on the asterisk box, then all of a sudden it just stopped working. I get the following errors (just keeps looping) *CLI DEBUG[1125329600]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found DEBUG[1125329600]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found DEBUG[1125329600]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Not Found DEBUG[1125329600]: File chan_sip.c, Line 4405 (handle_request): That's odd... Got a response on a call we dont know about. DEBUG[1125329600]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Not Found NOTICE[1125329600]: File chan_sip.c, Line 4233 (handle_response): Failed to authenticate on REGISTER to 'sip:[EMAIL PROTECTED];tag=as4b14216d' Any idea's what this could be? Is this an asterisk setup problem, or could it be FWD? Sorry to ask on this list, but I wasn't sure which one to ask on. Thanks, -- Leif Madsen - Telecommunications Technology ___ 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] Analog phone not ringing
--On Saturday, July 19, 2003 16:30:04 +0100 Darren Poulson [EMAIL PROTECTED] wrote: The one thing that I think it could be is the connector to convert from RJ45 to BT phone socket. I'm using a mod tap that I had lying around. Not sure what the wiring is like inside it. That's a pretty good bet. Depending on the phone I've regularly had to swap wires, solder wires together or add capacitors to get phones to work with my ATA 186 and even ordinary phone lines here in darkest Sussex. There seem to be a number of wiring variations that can cause problems - you can use google to get many references to the possible solutions. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7905G vs ATA186
As far as I know the Sip support for the 7905 has not been generally released so comments you've seen on this list refer to test versions of the code. You can set up a call between two phones on an ATA186 through asterisk. Iain --On Wednesday, July 16, 2003 9:28 pm +1000 Steven Honson [EMAIL PROTECTED] wrote: Hi All, I'm looking at getting some Cisco VoIP hardware to play with in combination with a Asterisk server. I've heard that there is beta software available to do SIP on the 7905G. So, I'm thinking of either getting a 7905G or a ATA186. My dillema is, which one to buy? I can get both for about the same price, has anyone had any experience with using a 7905G with Asterisk? On one hand it would be useful to have a ATA186 for its two ports, might be useful for testing stuff (Can you call between the two ports on a ATA186 ok?). But on the other hand, having a proper IP Phone would be cool also. Cheers, Steven ___ 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] wait and user input..
Not all of the * wait commands respond to dtmf whilst playing back. Couldn't you use the Background application to play the music? That does respond to dtmf whilst playback is in progress. Iain --On Friday, July 11, 2003 10:52 am + WipeOut . [EMAIL PROTECTED] wrote: Hi.. How do you accept user input while waiting or playing moh? My Dialplan is as follows.. ring,ring,.. Hello thanks for calling blah blah... Please enter the extention number blah blah... WaitMusicOnHold(10) If no input pass call to operator.. The problem is that the user has to input the extension while they are being told what to do.. any input during Wait or WaitMusicOnHold is ignored... Thanks.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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] SIP call transfers - any other way than using '#' ?
If you make an outgoing call to a conference bridge (or anything else that needs DTMF '#') then you can't use the asterisk 'T' transfer option because that is triggered by the '# also. Is there already a solution in # for this eg use two keys to trigger a transfer rather than just the '#'? Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD trouble - 407 error
Thanks for that. It seems one now needs something like this in sip.conf: [fwd.pulver.com] type=peer host=fwd.pulver.com username=12345 secret=mysecret fromdomain=fwd.pulver.com callerid=Free World Dialup All is well again ... Iain --On Saturday, July 5, 2003 9:31 pm -0400 James H. Cloos Jr. [EMAIL PROTECTED] wrote: Iain == Iain Stevenson [EMAIL PROTECTED] writes: Iain I didn't used to have any trouble with FWD and * is registering Iain with FWD OK. Has FWD changed or * changed in a way that might Iain cause this error? Jeff just announce an upgrade to fwd the other day. One change is that callers have to be logged in. (This is to help ensure the cnid info send in the sip invite accurately reflects the callers fwd number.) This change was added after a rash of nuisance calls?. -JimC ? As a disclaimer: I started a thread on the fwd list about these after I received a couple of mostly amusing ones; the discussion on that thread may have led to the change. ___ 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] FWD trouble - 407 error
I got this today trying to place a call through FWD: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 217.11.11.1:5060;branch=z9hG4bK230f856c From: Iain sip:[EMAIL PROTECTED];tag=as6eaa85fb To: sip:[EMAIL PROTECTED];tag=b27e1a1d33761e85846fc98f5f3a7e58.3701 I didn't used to have any trouble with FWD and * is registering with FWD OK. Has FWD changed or * changed in a way that might cause this error? Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question
RFC3389 is comfort noise. By default the ATA 186 will generate rfc3389 packets. You can turn this off through the ATA 186 web interface. It looks as though you need to configure that ATA186 properly - several people have posted guides on this. Iain --On Thursday, July 3, 2003 9:29 am + Andrey Katkov [EMAIL PROTECTED] wrote: Hi! I've installed Asterisk and connected ATA-186. When I press 8500, I listen voice main menu and prompt for enter mailbox number. I press 1234, but asterisk not accept number and switch to demo-instruct. Also Asterisk write warning: NOTICE[77839]: File rtp.c, Line 221 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible. Where am I wrong? -- Sincerely yours, Andrey Katkov. ___ 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] Problem with echo
rxgain and txgain are used, for example with the X100P. As I understand it, the echo problem with a SIP to PSTN implementation in * has two components: - echo resulting from the digital to analogue conversion at the X100P - acoustic feedback within the handset used The former is reduced by using the zaptel echo canceller set by this in zapata.conf: echocancel=yes echocancelwhenbridged=yes The choice of echo canceller to use is made when you compile zaptel. mec2 is the default. You can enable aggressive cancellation in mec2 but this can be a bit too severe making calls sound almost half duplex. Mec3 seems to be a bit unstable. You can reduce feedback related echo by tuning rxgain and/or txgain. A value of -3.0 will set the gain at about 70% of its initial value. Iain --On Wednesday, July 2, 2003 3:40 am -0700 Ing. Angel Gomez Garcia [EMAIL PROTECTED] wrote: I have a SIP FXO 8 port VoIP gateway, and it has a parameter called 'input gain' wich is the one I modified, there might be a similar parameter on the configuration for the hardware you are using. Dan wrote: Hi, What do you mean by pstn-gateway? There is no input gain parameter in zapata.conf file? It is about rxgain? BR, Dan - Original Message - From: Ing. Angel Gomez Garcia [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 11:48 AM Subject: Re: [Asterisk-Users] Problem with echo I had a similar problem and solved it changing the params of input gain on my pstn-gateway, change from a value of 10 to a value of 1 and that eliminated the echo on the SIP Phones. Dave Packham wrote: Same prob here. 15 SIP phones only get eco when going to the PSTN... if you find something let me know Dave [EMAIL PROTECTED] 7/1/2003 8:53:13 AM Hello, I can't have asterisk working without echo when I place a call from IP phone (SIP or H323) to a PSTN Phone. The called number as no problem with echo but there is a very audible echo in the SIP phone. This situation occurs either when connected with ISDN card thru i4linux driver and with my openline card from voicetronix. Do you have any suggestion fo that? Regards, Daniel ANDRE ___ 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More mec3 feedback
I had a call today where there were several remote participants using a speakerphone. They sounded quiet to me. Every time I spoke I got noise at my end but the respondents never complained of any problems hearing me. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mec3 - temporary call distortion
Whilst in a call using the mec3 echo canceller today I had period of about 20 seconds of speech distortion. It's hard to describe but to me the call sounded as though we were having the conversation in a bathroom with some extra noise bursts and echo thrown in. I could carry on the call, with difficulty, and my correspondent didn't complain of any noise at all. After that 20 seconds everything was fine again. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fixed point mec3
... but it still only works on x86? I get a failure to find asm/i387.h at line 69 of zaptel.c on my ppc box. Iain --On Sunday, June 29, 2003 11:55 am -0500 Mark Spencer [EMAIL PROTECTED] wrote: I've been working on a fixed point mec3 echo can. The old mec3 which crashed a lot of peoples machines because it was floating point in the kernel is now available as mec3-float.h. I'd appreciate any testing / feedback on the new fixed point mec3 echo can. Thanks! 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
Re: [Asterisk-Users] fixed point mec3
... thanks - seems to go now. I'll test some more. Iain --On Sunday, June 29, 2003 3:30 pm -0500 Mark Spencer [EMAIL PROTECTED] wrote: Oops, a remenent of when it was still FP. Should be fixed now. Mark On Sun, 29 Jun 2003, Iain Stevenson wrote: ... but it still only works on x86? I get a failure to find asm/i387.h at line 69 of zaptel.c on my ppc box. Iain --On Sunday, June 29, 2003 11:55 am -0500 Mark Spencer [EMAIL PROTECTED] wrote: I've been working on a fixed point mec3 echo can. The old mec3 which crashed a lot of peoples machines because it was floating point in the kernel is now available as mec3-float.h. I'd appreciate any testing / feedback on the new fixed point mec3 echo can. Thanks! 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P and PSTN caller id
I think the problem is more fundamental than this. The state machine in the X100P assumes that nothing at all happens before a ring - so it will simply ignore everything (eg UK caller ID tones) until it gets that first ring to wake it up. Handling UK caller ID needs a re-write of the X100P driver. Iain --On Thursday, June 26, 2003 3:59 pm +0300 Dan [EMAIL PROTECTED] wrote: Gary sais some time ago: Have you got the number of rings for caller-id set right ?? its in chan_zap somewhere from memory.. It seems that in US it is after first ring, but in Europe it is before the first ring (but not allways). Anybody knows where this parameter is in chan_zap.c file? As for me it swhos my number, then it reads it, but not the right parameter. Thanks, Dan - Original Message - From: K. C. Li [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 26, 2003 3:15 PM Subject: Re: [Asterisk-Users] X100P and PSTN caller id On Thu, 26 Jun 2003, Andy Powell wrote: I don;'t know what else to try, I've had callerid turned on here but it doesn't work at all... :( The Call ID function also doesn't work in the UK. Regards, Kwong Li [EMAIL PROTECTED] Laser Business Systems Ltd. http://www.laser.com ___ 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] PHP MySQL cdr interface?
Roy Sigurd Karlsbakk posted a php utility to calculate call costs to this list a while back. I hacked it for my own use and you can have that if you'd like to improve it/make it general purpose. Iain --On Tuesday, June 24, 2003 11:18 am -0400 Marcus Adolfsson [EMAIL PROTECTED] wrote: Before I reinvent-the-wheel, does any one know of PHP based interface to the CDR table? If not, I will get started writing one. Thanks, Marcus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asteisk, sip NAT
--On Sunday, June 22, 2003 14:38:20 +0200 Hervé Thibaud [EMAIL PROTECTED] wrote: i have an error when i start asterisk in : chan_modem.so (Generic Voice Modem Driver) -- Parsing /etc/asterisk/modem.conf': Found -- Loading modem driver chan_modem_i4l.so = (ISDN4Linux Emulates Modem Driver) Warning(32771): File chan_oss.c Line 228 (sound_thread): Read error on sound device; Ressource temporarily unavilable Probably means some other program has already locked the sound output on your * box. You can put: noload = chan_oss.so in /etc/asterisk/modules.conf if you don't need OSS sound for *. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] best ISDN BRI solution for DID
--On Saturday, June 21, 2003 06:28:32 + WipeOut . [EMAIL PROTECTED] wrote: So far I have just got it to the point where I am able to make calls and have not had the serious echo problems that everyone warns about when using a passive card.. . ... you're using chan_capi - maybe that's the answer to the echo issue. My experience wirh echo and delay problems was with a passive card on i4l. IMHO new users need to be careful to pick the right ISDN card or they'll end up disappointed. With some cards there is also the issue of capi drivers ie make sure there is one for your chosen Linux platform before you buy the card. I may be the only ppc user on the list but you can reckon any binary capi driver will only be x86. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Poor quality with FWD - codec selection issue?
A colleague called me through my * system via FWD using SJPhone and the quality was distinctly poor - a lot of hum and delay. Looking at the debug log the codec used was miscellaneously numbered 0, 4 and 8. I thought I'd disabled 4 (g.723) but it appears not. My sip.conf has this: general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = voip-sip defaultexpiry = 3600 register = 12345:[EMAIL PROTECTED]/39 disallow=all allow=alaw allow=ulaw I was expecting this would stop g.723 from being even tried - am I missing something? Is there any config option for SJphone that clobbers g.723? Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] i4l - summary of patches?
--On Thursday, June 19, 2003 17:24:21 +1000 Adam Goryachev [EMAIL PROTECTED] wrote: One problem I had with this problem is when I dial out through asterisk, once I have dialled, the remote end doesn't detect my dtmf key presses. ie, I can diall (eg a bank) but when they ask to press 3 for assistance, I can press 3 many times, but they never realise I have pressed it. Any ideas on how to resolve this? Have you installed the asterisk dsp patch for i4l? It's so long since I ran ISDN that I can't remember for sure whether DTMF was handled correctly but I think it was. I was using 2 patches - the one to disable kernel DTMF and silence suppression and Pauline's DSP patch for chan_modemi4l. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN4Linux + Asterisk and Europe
There has been a lot of discussion about ISDN BRI on the list - a search will turn up plenty of discussion! You're right about there being a lot of ISDN cards available that are certified for use in Europe. They fall into two categories - active and passive. Passive cards are cheap and generally operate through ISDN4linux and asterisk's chan_modemi4l. The reported disadvantages of this approach are: - delay during calls - echo (it's disputed what the cause of it is but it's a bit of a nuisance) - call tones don't follow PSTN patterns. The active cards (AVM is the major supplier, I think) are better. If you get one with a CAPI interface then you can use the asterisk chan_capi driver. I haven't any experience of this type of card - maybe someone else can provide feedback. Iain --On Monday, June 2, 2003 12:33 am +0200 Piotr Adamiak [EMAIL PROTECTED] wrote: Hello, Anyone on this group using / implementing * and hardware certified for use in Europe ? I believe that ISDN4Linux cards mostly have telecomm certificates, so using them should be safe on the client side. Are there any major issues / problems associated with using such cards with * ? I am talking about a small / very small office with single - few lines. All the best, Piotr Adamiak -- I am updating my DNA. 'Microsoft Genome 2.0' says I need to reboot my body to continue. I am worried that I may not have enough free chromosomes to allow the full installation to complete. ___ 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] CDR ??
Don't forget to set the database permissions. These need to agree with whatever is in /etc/asterisk/cdr_mysql.conf. Iain --On Friday, March 28, 2003 3:14 pm + WipeOut . [EMAIL PROTECTED] wrote: Hi, I see in /ect/asterisk there is a cdr_mysql.conf to configure the CDR logging to a MySQL DB.. I have a couple of questions.. 1. Where do I find the DB schema to create the DB? (may be a good idea to add this to the top of the .conf file in the cvs so that it is easy to find for amyone wanting to set it up.. Just a thought.) 2. Are there any req's to making this work? (apart from, I assume, having the mysql client installed on the * box..) I also read in the archives that CDR logging could be done to a CSV file, How and where is this setup and configured? Thanks.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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] Linux Kernel Patch
--On Monday, March 24, 2003 1:00 pm +1100 Adam Goryachev
[EMAIL PROTECTED] wrote:
Does anyone know the location of the kernel patch to disable isdn dtmf
detection?
The patch below should do that.
Also the location of the asterisk patch for doing the dtmf detection?
Pauline Middelink posted it to the list - search on her name and it'll turn
up.
Iain
--- /build/linux-2.4.20/drivers/isdn/isdn_tty.c Fri Dec 21 17:41:54 2001
+++ isdn_tty.c Sat Feb 1 09:14:33 2003
@@ -133,9 +133,9 @@
if (info-online) {
r = 0;
#ifdef CONFIG_ISDN_AUDIO
- isdn_audio_eval_dtmf(info);
- if ((info-vonline 1)
(info-emu.vpar[1]))
- isdn_audio_eval_silence(info);
+// isdn_audio_eval_dtmf(info);
+// if ((info-vonline 1)
(info-emu.vpar[1]))
+// isdn_audio_eval_silence(info);
#endif
if ((tty = info-tty)) {
if (info-mcr UART_MCR_RTS) {
@@ -190,10 +190,10 @@
#ifdef CONFIG_ISDN_AUDIO
ifmt = 1;
- if ((info-vonline) (!info-emu.vpar[4]))
- isdn_audio_calc_dtmf(info, skb-data, skb-len, ifmt);
- if ((info-vonline 1) (info-emu.vpar[1]))
- isdn_audio_calc_silence(info, skb-data, skb-len, ifmt);
+// if ((info-vonline) (!info-emu.vpar[4]))
+// isdn_audio_calc_dtmf(info, skb-data, skb-len, ifmt);
+// if ((info-vonline 1) (info-emu.vpar[1]))
+// isdn_audio_calc_silence(info, skb-data, skb-len, ifmt);
#endif
if ((info-online 2)
#ifdef CONFIG_ISDN_AUDIO
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Re: [Asterisk-Users] X101P minor nuisances..
Do you know for sure whether the PBX issues a call termination pulse (ie zero or reverse battery) on completion of a call? Iain --On Friday, March 21, 2003 8:56 pm +0100 Florian Overkamp [EMAIL PROTECTED] wrote: Hi guys, So, now I've made a small demo box to do some IVR apps and hooked it up to an analog line of an Ericsson MD110 pbx. Everything seems to work fine, but: issue: even though X101P is configured for kewlstart it fails to see disconnects unless I enable busydetect issue: i don't get callerid, even though the pbx techs ensure me it is on the line (so, disconnect supervision and callerid seem broken) This may well be my configuration, but please tell me where to do what. Thanks ! Here is what I've got now: /etc/zaptel.conf: # X100P fxsks=1 # zones loadzone = nl defaultzone=nl /etc/asterisk/zapata.conf [channels] language=nl usecallerid=yes transfer=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 ; seems nessecary ? busydetect=yes busycount=5 ; Individual channels context=default signalling=fxs_ks channel = 1 ___ 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] inbound isdn call
Sounds like the i4l dtmf problem. Assuming you are using i4l, the kernel dtmf detection routines are poor and quite frequently misinterpret speech as dtmf tones. You need to patch asterisk to handle dtmf and i4l not to detect dtmf (or silence). There are a few posts on this list about fixing this issue. Iain --On Thursday, February 27, 2003 12:06 pm +0100 Marian Danisek [EMAIL PROTECTED] wrote: hello, when call is made via asterisk from isdn line to the snom sip phones and caller on isdn line is speaking loudly to the microfone, people on the sip phones didnt hear voice but tones, like dtmf. how can i firuge out this problem ? can echo cancel algorithm ? best regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ 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] inbound isdn call
--On Thursday, February 27, 2003 1:54 pm +0100 Marian Danisek [EMAIL PROTECTED] wrote: this mean that i need 2 different patches ? I already found isdn_audio.c and isdn_audio.h patch... this is for i4l. You meat that i need another patch for asterisk ? If you want asterisk to handle dtmf then you need Pauline Middelink's dsp patch - isdn-dsp.txt - which was posted to the list in January. This patch allows asterisk's dsp routines - the same ones used for the zaptel interfaces - to provide dtmf support for the ISDN line. Without it, you will have no dtmf support if you apply the isdn-audio.c patch. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
