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 stati

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

Re: [Asterisk-Users] Park and out-going trunk calls.

--On Wednesday, August 13, 2003 1:58 pm -0500 James Sizemore <[EMAIL PROTECTED]> wrote: If you add "t" to you out-going trunk "Dial" lines: exten => _NXX,1,Dial(SIP/[EMAIL PROTECTED]||t) exten => _NXX,2,Congestion so that you can still use park to park a call or transfer the phones, Yo

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 correctl

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 U

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

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

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

[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 re

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 Iain

RE: [Asterisk-Users] RTP codec 13 received - Ciscoincompatibilit y?

.. poking head above parapet, venturing correction .. RTP payload type 13 is "comfort noise" viz whereas payload type 19 is "reserved". Maybe Cisco is right ;-) I believe * has a partial implementation of comfort noise but that it's not complet

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 i

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]

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 - chan_capi is reportedly the best performing ISDN channel driver for asterisk, although I personally haven't used it ;-) Iain --On Sunda

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

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

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 <[EM

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 y

[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 '#'?

Re: [Asterisk-Users] FWD trouble - 407 error

;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 e

[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" ;tag=as6eaa85fb To: ;tag=b27e1a1d33761e85846fc98f5f3a7e58.3701 I didn't used to have any trouble with FWD and * is registering wit

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 +

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 usin

[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

[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

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

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 peopl

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 d

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: Be

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)

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 th

[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] p

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 ass

Re: [Asterisk-Users] i4l - summary of patches?

You probably want to remove the i4l handling of DTMF and silence suppression. You can do this by commenting out the signal processing routines in the kernel i4l code. This stops wasted work from being done. I posted the patch below a while back - not sure it it still works. Iain --- /buil

Re: [Asterisk-Users] Quicknet.-

--On Saturday, June 14, 2003 12:56:09 -0400 Francisco Perez-Landaeta <[EMAIL PROTECTED]> wrote: Hi, I am sure this question has been asked a hundred times. Yes - search the list. I am planning on purchasing the Dev Kit to experience with * and do some voip configurations. However, I see that Q

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 genera

[Asterisk-Users] SIP autodestruct bug affects MOH too

I built the cvs as of last night and found that a call from a SIP terminal (ATA 186) to music on hold bombed thus after a few seconds: -- Executing Answer("SIP/cisco1-e146", "") in new stack -- Executing MP3Player("SIP/cisco1-e146", "/var/lib/asterisk/mohmp3/test.mp3") in new stack WARNING[6

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

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

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

Re: [Asterisk-Users] X100P in Europe/Austria

The preceding comments probably apply to direct analogue PSTN connections. You may have problems if the line you are connecting to is from a PBX or ISDN terminal adapter. Iain --On Thursday, March 13, 2003 11:49 am +0100 Klaus Darilion <[EMAIL PROTECTED]> wrote: Hello! Does the X100P card

Re: [Asterisk-Users] ATA186 (was Windows XP client?)

--On Saturday, March 8, 2003 5:12 pm -0500 Jim Archer <[EMAIL PROTECTED]> wrote: The Cisco ATA 186 is installed at the subscriber`s premises and supports two voice ports, each with its own independent phone number. But there seemed to be something about the licensing that prohibited the use of

[Asterisk-Users] SIP rings on after voicemail answers

I have asterisk set up to ring a local SIP phone (on an ATA186) for incoming calls and to divert to voicemail after 20 seconds. However, when the 20 seconds is up, asterisk answers (I know this because I have another phone on the same line as asterisk) but the SIP phone keeps on ringing for a

Re: [Asterisk-Users] Asterisk & X100P in Finland?

--On Thursday, March 6, 2003 12:15 am +0200 Juha Suhonen <[EMAIL PROTECTED]> wrote: I'm having problems with Asterisk and Digium's X100P (using kewlstart) in Finland - Asterisk doesn't detect POTS hangup. Do you mean that asterisk doesn't detect the end of an incoming POTS call when the callin

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

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

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