Re: [asterisk-users] [Fwd: confirm f1ab6c493110edited]
Doug Lytle wrote: Your membership in the mailing list asterisk-users has been disabled due to excessive bounces The last bounce received from you was dated Anybody else seeing this? My mail server logs don't show any issues. Doug I just did yes, Don't know why :) Dunc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
Gordon Henderson wrote: On Sat, 23 May 2009, Dunc wrote: Hi Gordon, thanks for your reply, I was hoping to find someone who was involved in the thread before. See inline comments Hi everyone, I just found this thread, which is amazing as I'm on my first go with asterisk and so far I've been pulling my hair out for the last week :-) I have 2 questions which were raised while this fault was being debugged. 1) Gordon says:- Is this a place where you get a polarity reversal event on call startup? In the UK we do. (Well on BT lines - I've a funny feeling some Telewest/NTL lines use Bell signaling). On an incoming call we get: Polarity reversal. FSK Caller ID burst Ringing Well I've got an NTL phone line, can anyone tell me what to use for that? Throw it away and get a BT one ;-) I'd suggest running with verbose =3 and seeing what it says. However I have 2 clients with Teleworst lines and I've never been able to make caller ID work on them, even though teleworst insist they are providing caller ID.. Well that's a shame because I specifically went for NTL so that I haven't got BT to contend with when my Internet is broken. The Ethernet presentation instead of a pair of BT wires is very appealing. I'll be happy to get it up and running for now and worry about caller ID later though. So we'll press on :-) 2) Do I still need the same 2pin cable? Because I've been to Maplins too and bought one that I thought was right, but this one is a 4pin too. Can anyone tell me which pins on their 2pin cable are connected at each end? I'll bodge my cable until it works and then get a proper one once I'm sure. I think part of this same thread had something about modem vs. ordinary cables - however I put in a 4-pin modem cable to see what happenes and it continued to work as before, so I'm personally not convinced about that one... ie. I've not had a cable that didn't work. Gordon Right so I'm probably barking up the wrong red herring with the cable problem. I did wonder when the TDM card has 2 pins on the RJ11 socket what possible difference a 4 pin cable could make, but I've definitely had madness with phones before where wiring up seemingly unused wires made the extensions ring, so I'm willing to accept phone wiring is mad. In that case, I'll reply to an earlier post in the thread from someone requesting my config / symptoms, and we'll take it from there. Thanks for your help. Cheers, Dunc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
that works now (please correct me if I'm wrong anyone) so hopefully the stuff below is irrelevant now. Thanks in advance to anyone who makes it to here :-) and I hope someone can point me in the right direction. Cheers, Dunc it seems that a 2-pin cable from the wall socket to the card is required. Now, I was warned about cables and did my best to get the one that sounded right, however mine definitely has 4 pins. So my 2 questions are 1) What are the pinouts for the 2 pin cable, and I'll make my own for now (For bonus points, unless the wires are crossed over, what possible difference could it make when the TDM card only has 2 pins anyway?) 2) Is this the correct cable for NTL too? This line that you're using, can you use a regular analog phone to make call through it? I don't have any server running in UK, only USA and Latin America... But I'll assume the cabling is the same... If so you should have a connector like this: http://img.zdnet.com/techDirectory/RJ11.GIF Test your line with a regular phone. Make sure it works fine. Also make sure not to connect the PSTN line on the FXS card, you can 'burn' your FXS doing that. I think I should find out definite answers to the above before I worry any further about the card and Asterisk :-) I agree... =) Thanks for getting back to me, hope you can help. No prob, I hope too! Cheers! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
Tzafrir Cohen wrote: On Sun, May 24, 2009 at 12:36:26PM +0100, Dunc wrote: Hi Tiago, I have an OpenVox A400P11, it shows up like this... eddie ~ # lsdahdi ### Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) 1 FXSFXSKS (EC: MG2) RED 2 FXOFXOLS (EC: MG2) Something is wrong here. FXS channels should have FXO signalling and vice versa. The cause for that is probably a bug in dahdi_genconf (or rather: in Dahdi::Chan) in 2.1.0 (fixed shortly after the release) that misdetected the channels and thus generated a wrong dahdi_channels.conf and still misleads you here. If you'll switch to 2.2 the problem will go away :-( (Evidently system.conf has the right signalling, otherwise it wouldn't have applied). Hmm, my channel 1 is the FXO (i.e. connects to the wall socket) and channel 2 is FXS (that's where I plugged my phone line in) Use of uninitialized value in string eq at /usr/lib64/perl5/site_perl/5.8.8/Dahdi/Chans.pm line 221. 3 unknown Use of uninitialized value in string eq at /usr/lib64/perl5/site_perl/5.8.8/Dahdi/Chans.pm line 221. 4 unknown eddie ~ # I'm pretty sure that the RED alarm is a bad thing. While googling about this error from the asterisk console *CLI [May 23 18:14:02] NOTICE[4469]: chan_dahdi.c:8164 handle_init_event: Alarm cleared on channel 1 [May 23 18:14:03] WARNING[4469]: chan_dahdi.c:4664 handle_alarms: Detected alarm on channel 1: Red Alarm I discovered this thread on the mailing list, and so signed up and mailed in with the same subject. It didn't link them together though it seems, so here's a URL http://www.mail-archive.com/asterisk-users@lists.digium.com/msg223429.html If you read this specific post, and the last few before http://www.mail-archive.com/asterisk-users@lists.digium.com/msg223523.html How is your configuration? Please post here. At first I thought you were using a A1200P, this card (which is very good) gave some headache some weeks ago, because it wasn't working properly with dahdi. I didn't have a chance to test it again. It's because for this card you need a module from Openvox. The A400P card that they made is a clone of the TDM and should work fine with DAHDI. For this card you shouldn't need any module from Openvox. Hi, My config is mostly out-of-the-box apart from where I've needed to tweak things. This is my first attempt with Asterisk though, but I've been playing on and off for a couple of weeks now. I'm pretty sure that I've got DAHDI set up correctly as you can see in lsdahdi above. Channel 1 connects to my PSTN and Channel 2 has an analogue phone connected. I can definitely make calls with the analogue phone connected straight into the PSTN to answer your question below. I'd still like to know what the RED alarm means if anyone can tell me though... For an FXO channel: not connected. Or rather: not detecting battery current from the remote FXS. I have also definitely got my wctdm loaded in UK mode eddie etc # dmesg | grep -i fx Module 0: Installed -- AUTO FXO (UK mode) Module 1: Installed -- AUTO FXS/DPO Here is my dahdi/system.conf # Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) fxsks=1 echocanceller=mg2,1 fxols=2 echocanceller=mg2,2 # channel 3, WCTDM/4/2, no module. # channel 4, WCTDM/4/3, no module. # Global data loadzone = uk defaultzone = uk Ok, on to asterisk. The tweaks I have made to the defaults are:- chan_dahdi.conf --- cidsignalling=v23 sendcalleridafter = 2 And I have #included an extra file like this. #include /etc/asterisk/dahdi-channels.conf which contains eddie asterisk # cat /etc/asterisk/dahdi-channels.conf ; Autogenerated by /usr/sbin/dahdi_genconf on Sat May 23 14:19:49 2009 -- do not hand edit ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/asterisk/chan_dahdi.conf that will include the global settings ; ; Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) ;;; line=1 WCTDM/4/0 signalling=fxs_ks signalling=fxo_ks callerid=asreceived group=0 context=from-pstn channel = 1 callerid= group= context=default ;;; line=2 WCTDM/4/1 signalling=fxo_ls signalling=fxs_ls callerid=Channel 2 4002 mailbox=4002 group=5 context=from-internal channel = 2 callerid= mailbox= group= context=default After those modifications, re-run dahdi-cfg and in Asterisk: asterisk -rx 'dahdi restart' Are you absolutely sure this is the right way round. I thought that an FXO module connected to the wall, and used FXS signalling. And vice versa. My channel 1 is the one connected to PSTN so surely it should have FXS signalling? Please excuse my ignorance, I don't mean to be telling you what to do, it's just that this is different to what I've read everywhere else
Re: [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
Gordon Henderson wrote: On Sun, 24 May 2009, Dunc wrote: extensions.conf --- And the same for extensions.conf.. eddie asterisk # cat /etc/asterisk/extensions_dunc.conf [from-sip] exten = 07875123456,1,Dial(DAHDI/1/07875123456,20) [from-internal] exten = 07875123456,1,Dial(DAHDI/1/07875123456,20) [from-pstn] exten = s,1,Dial(DAHDI/2/,20) Start simpler: [from-pstn] exten = s,1,Answer() exten = s,n,Playback(demo-congrats) exten = s,n,Hangup() Same for the analogue phone. Gordon Hi again Gordon, As suggested. eddie ~ # cat !$ cat /etc/asterisk/extensions_dunc.conf [from-internal] exten = 123456,1,Answer() exten = 123456,2,Playback(demo-congrats) exten = 123456,3,Hangup() [from-pstn] exten = s,1,Answer() exten = s,2,Playback(demo-congrats) exten = s,3,Hangup() If I dial 123456 on my analogue phone, then it's all cool. *CLI -- Starting simple switch on 'DAHDI/2-1' -- Executing [123...@from-internal:1] Answer(DAHDI/2-1, ) in new stack -- Executing [123...@from-internal:2] Playback(DAHDI/2-1, demo-congrats) in new stack -- DAHDI/2-1 Playing 'demo-congrats.gsm' (language 'en') -- Executing [123...@from-internal:3] Hangup(DAHDI/2-1, ) in new stack == Spawn extension (from-internal, 123456, 3) exited non-zero on 'DAHDI/2-1' -- Hungup 'DAHDI/2-1' *CLI (I'm not actually sure that it should exit non-zero or not, but I definitely got the recorded message and then it hung up) So that's fine. There's just incoming to fix now. When I dial in from external:- *CLI [May 24 15:16:56] NOTICE[3834]: chan_dahdi.c:8164 handle_init_event: Alarm cleared on channel 1 -- Starting simple switch on 'DAHDI/1-1' [May 24 15:16:56] NOTICE[3889]: chan_dahdi.c:7505 ss_thread: Got event 18 (Ring Begin)... [May 24 15:16:56] NOTICE[3889]: chan_dahdi.c:7505 ss_thread: Got event 2 (Ring/Answered)... [May 24 15:16:57] NOTICE[3889]: chan_dahdi.c:7505 ss_thread: Got event 4 (Alarm)... -- Executing [...@from-pstn:1] Answer(DAHDI/1-1, ) in new stack -- Executing [...@from-pstn:2] Playback(DAHDI/1-1, demo-congrats) in new stack -- DAHDI/1-1 Playing 'demo-congrats.gsm' (language 'en') -- Executing [...@from-pstn:3] Hangup(DAHDI/1-1, ) in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' *CLI Looks good, but my incoming call from mobile doesn't actually stop ringing, I still end up at Virgin's voicemail. I notice that the first line is an alarm clear event. So it looks like my RED alarm is on right up until there's an incoming call. Other people on this thread have said that RED means can't detect a voltage from the line. So that's sort of making sense I think, as there'll definitely be a voltage I've just tried your suggestion from your other mail, where you were on about using the cable from a phone. So I pinched the one from my analogue phone (it's 2 wires.) The alarm cleared as soon as plugged it in, and now everything is working :-) So it WAS the cable all along! Who'd have thought. I've just tried using the cable I was using between the card and the wall on my phone, and it doesn't work. I can't believe I didn't think to try swapping them before. Now I need to hunt down another one of those cables then. Thanks so much to everyone who has helped me with this, it's much appreciated. Now I can get on with playing with asterisk. Cheers, Dunc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
Hi everyone, I just found this thread, which is amazing as I'm on my first go with asterisk and so far I've been pulling my hair out for the last week :-) I have 2 questions which were raised while this fault was being debugged. 1) Gordon says:- Is this a place where you get a polarity reversal event on call startup? In the UK we do. (Well on BT lines - I've a funny feeling some Telewest/NTL lines use Bell signaling). On an incoming call we get: Polarity reversal. FSK Caller ID burst Ringing Well I've got an NTL phone line, can anyone tell me what to use for that? 2) Do I still need the same 2pin cable? Because I've been to Maplins too and bought one that I thought was right, but this one is a 4pin too. Can anyone tell me which pins on their 2pin cable are connected at each end? I'll bodge my cable until it works and then get a proper one once I'm sure. Thanks in advance. Dunc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
Tzafrir Cohen wrote: On Sat, May 23, 2009 at 04:02:57PM +0100, Dunc wrote: In the UK we do. (Well on BT lines - I've a funny feeling some Telewest/NTL lines use Bell signaling). On an incoming call we get: Polarity reversal. FSK Caller ID burst Ringing https://issues.asterisk.org/view.php?id=9096 ? Thanks for the link, at the moment my stuff doesn't work properly at all though, it's not just the caller ID stuff. I'm guessing it's down to the cable though now after reading the previous posts on this thread. If someone knows exactly which pins at each end I need to connect with a 2 wire cable that would be amazing (UK NTL phone line.) (Bueller, anyone? :-) ) Once I have incoming calls working then I think I'll be back fixing the caller ID stuff :) Cheers, Dunc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [UK SPECIFIC] DAHDI and a OpenVox Card
Tiago Durante wrote: Dunc On Sat, May 23, 2009 at 12:43 PM, Dunc d...@lemonia.org wrote: Tzafrir Cohen wrote: On Sat, May 23, 2009 at 04:02:57PM +0100, Dunc wrote: In the UK we do. (Well on BT lines - I've a funny feeling some Telewest/NTL lines use Bell signaling). On an incoming call we get: Polarity reversal. FSK Caller ID burst Ringing https://issues.asterisk.org/view.php?id=9096 ? Thanks for the link, at the moment my stuff doesn't work properly at all though, it's not just the caller ID stuff. I'm guessing it's down to the cable though now after reading the previous posts on this thread. If someone knows exactly which pins at each end I need to connect with a 2 wire cable that would be amazing (UK NTL phone line.) (Bueller, anyone? :-) ) Once I have incoming calls working then I think I'll be back fixing the caller ID stuff :) I've some experience with openvox cards, what card are you using? What problems are you having? Cheers, Hi Tiago, I have an OpenVox A400P11, it shows up like this... eddie ~ # lsdahdi ### Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) 1 FXSFXSKS (EC: MG2) RED 2 FXOFXOLS (EC: MG2) Use of uninitialized value in string eq at /usr/lib64/perl5/site_perl/5.8.8/Dahdi/Chans.pm line 221. 3 unknown Use of uninitialized value in string eq at /usr/lib64/perl5/site_perl/5.8.8/Dahdi/Chans.pm line 221. 4 unknown eddie ~ # I'm pretty sure that the RED alarm is a bad thing. While googling about this error from the asterisk console *CLI [May 23 18:14:02] NOTICE[4469]: chan_dahdi.c:8164 handle_init_event: Alarm cleared on channel 1 [May 23 18:14:03] WARNING[4469]: chan_dahdi.c:4664 handle_alarms: Detected alarm on channel 1: Red Alarm I discovered this thread on the mailing list, and so signed up and mailed in with the same subject. It didn't link them together though it seems, so here's a URL http://www.mail-archive.com/asterisk-users@lists.digium.com/msg223429.html If you read this specific post, and the last few before http://www.mail-archive.com/asterisk-users@lists.digium.com/msg223523.html it seems that a 2-pin cable from the wall socket to the card is required. Now, I was warned about cables and did my best to get the one that sounded right, however mine definitely has 4 pins. So my 2 questions are 1) What are the pinouts for the 2 pin cable, and I'll make my own for now (For bonus points, unless the wires are crossed over, what possible difference could it make when the TDM card only has 2 pins anyway?) 2) Is this the correct cable for NTL too? I think I should find out definite answers to the above before I worry any further about the card and Asterisk :-) Thanks for getting back to me, hope you can help. Cheers, Dunc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] External MW Lamp On/Off
Greg,
Yes, it helps quite a bit. It shows me where Comedian Mail spawns the
external app.
Do you have a copy of your SIP MWI script? I may be able to use it as a
starting point.
FWIW, I've been using my extensions.conf to set/unset MWI on phones
attached to Cisco Call Manager - it's a bit of a hack but I couldn't
find anything better searching around. We've got CM4 interconnected to
* with a SIP trunk.
To change the MWI status I spoof the caller ID and send a call from * to
the CM set or unset number, which doesn't sound so different from what
you need to do other than it's a SIP call that changes the MWI. I guess
you wouldn't need to worry about caller id as you'd be dialling out on
an analogue line.
It's not very pretty, but it seems to work OK, the main drawback is that
if a user retrieves their mail from someone else's phone the light stays
lit. I think that's fixable, but at the moment it's not a big deal for me.
Extract below - 100 is the voicemail entry point, and the 600/700
sequences in the h extension deal with figuring out what to do after
exit from voicemail.
Dunc
---
[globals]
VMAIL=0
[local]
; h - hangup
;
exten = h,1,GotoIf($[${VMAIL} != 0]?600)
exten = h,2,Hangup
;
; When exiting voicemail, check for new messages in the recipients
; mailbox and check that their MWI is set accordingly. [EMAIL PROTECTED]
; unsets MWI, [EMAIL PROTECTED] sets. Silly numbers that came about from
; getting the config togther. They need changing.
;
; This stuff actually needs to be in a context of its own, so that
; the h extension doesn't have to have the gotoif stuff. (maybe)
exten = h,600,SetCIDNum(${VMAIL})
exten = h,601,SetGlobalVar(VMAIL=0)
exten = h,602,HasNewVoicemail([EMAIL PROTECTED]:INBOX)
exten = h,603,Dial(SIP/[EMAIL PROTECTED])
exten = h,604,Hangup
exten = h,703,Dial(SIP/[EMAIL PROTECTED])
exten = h,704,Hangup
;
;
;
; Voicemail.
; First, check if the call is a redirection (ie someone
; being transferred in to leave a message) - CM redirects to 1+ccm ext
; to indicate that this is the case. Set $VMAIL to the destination
; mailbox for exit handling (ugly).
;
; If it's not a redirect, send to voicemail with the callerid as the
; mailbox, otherwise use the diversion field.
exten = 100,1,Wait(1)
exten = 100,2,GotoIf($[${RDNIS}:1]?9)
exten = 100,3,SetGlobalVar(VMAIL=${CALLERIDNUM})
exten = 100,4,VoicemailMain(${CALLERIDNUM})
exten = 100,5,Hangup
exten = 100,9,SetGlobalVar(VMAIL=${RDNIS:1})
exten = 100,10,Voicemail(u${RDNIS:1})
exten = 100,11,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
