similar to: DAHDI - analogue, not seeing ringing (UK)

Can anyone tell me what this message means? Got event 17 (Polarity Reversal)... I'm running DAHDI 2.0 with a TDM401 card. Asterisk version 1.6.0. It appears that I get this Polarity Reversal each time an inbound call hangs up. This results in another ring, but no one is there. It appears as an unknown caller, but I believe its a phantom. Thanks, Jim [Oct 10 12:47:54] NOTICE[6669]:

Hi list, I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS 5. I am having trouble to make my simple IVR extension work, here is relevant config: zapata.conf ---- context=incoming signalling=fxs_ks channel => 4 context=internal signalling=fxo_ks channel => 1 ----- extensions.conf: ---- [office] exten => s,1,Dial(Zap/1,30) [home] exten =>

Hi, I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm having problems with Caller ID. I have run clidtest, and it seems happy enough, returning:- server clidtest # ./clidtest /dev/zap/1 Number: 0412222222, Name: MOBILE (that number's fake.) However, I'm not getting the caller ID passed through with *. Sometimes I get a "failed checksum" like

Howdy, My first forray into using res_mysql.conf for realtime access of sip users and extensions. I have the following relevant section of extensions.conf: --- [trunklocal] exten => _NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [local] include => trunklocal include => trunktollfree [longdistance] include => local include => trunkld [international] include

hi all, We have a TDM400 card with 4 wfo modules. now the modules load fine and when i start asterisk with on phone line connected it just starts spewing these messages: -- Starting simple switch on 'Zap/4-1' Jan 13 12:59:47 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)... Jan 13 12:59:47 NOTICE[10831]: chan_zap.c:5367 ss_thread: Got event 2 (Ring/Answered)...

Using the latest SVN of zaptel and asterisk, I can no longer receive incoming analog calls. The caller just hears it ringing but Asterisk doesn't pick up. I am seeing these error messages: [Apr 9 09:38:02] WARNING[16564]: chan_zap.c:6470 ss_thread: CallerID returned with error on channel 'Zap/2-1' [Apr 9 09:38:50] WARNING[16570]: chan_zap.c:6470 ss_thread: CallerID

Aloha, I'm finishing up the final touches on this install, and have run into an odd problem. I can't seem to get Caller ID on the analog phone lines working. It's a Digium AEX 410 card. I have Verbose set and a line to print the CID: I have usecallerid=yes and callerid=asreceived set in both chan_dahdi.conf and users.conf [analog] include=>default exten =>

pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4

Sorry if this is a daft question but when a PSTN call comes in on my X100P the console shows the following; NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2 (Ring/Answered)... NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2 (Ring/Answered)... NOTICE[1217602880]: File chan_zap.c, Line 4456 (ss_thread): Got event 2 (Ring/Answered)...

I am a little green when it comes to all this but I am trying to connect our PBX to an asterisk server using a TDM400 with 4 FXO modules. I am able to dial an extension on my PBX handset and I get a dialtone from the PBX. After 2 rings I then hear the asterisk server connect and I get a dialtone from asterisk. I am then able to dial an extension on another asterisk server. My question is: How do

Hi ! I ma having trouble with my PTSN line. When I call to my asterisk I get this.. -- Executing [s at from-pstn:3] Hangup("Zap/5-1", "") in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' -- Starting simple switch on 'Zap/5-1'[Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17

Hi all, I have hit a wall configuring a TDM400, I have set these up before without issue but today I just can't seem to figure out what I am doing wrong. On an incoming call the following is produced in the Asterisk console with verbose 4 -- Starting simple switch on 'Zap/2-1' Mar 22 16:12:34 NOTICE[2051]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... Mar 22

Hi all I?m a newbie to asterisk and I have install and configure asterisk 1.4.5 I have made some test and have face a strange behaviour I hava a simple dialplan when a call is receive from PTSN, [PSTN] exten => s,1,Answer() exten => s,2,Playback(intro-sicx) ; Listen to your voice exten => s,3,Dial(SIP/steph) exten => s,4,Hangup() I got the following when a call is

Hello all. This is my very first message to the list. I have a TDM400P card, It has 2 FXO channels which are connected to extensions of my PBX (Ericsson BP250), so I can dial from any SIP softphone directly to physical (analog and digital) extensions on my company. My PBX is configured so when I dial 8 on any extension, it will redirect to the first free FXO channel on my TDM400P card.

hi, when i call asterisk (on x100p) i got this : CLI> -- Starting simple switch on 'Zap/7-1' Jul 13 15:03:34 ERROR[311316]: callerid.c:192 callerid_feed: fsk_serie made mylen < 0 (-9) Jul 13 15:03:34 WARNING[311316]: chan_zap.c:4735 ss_thread: CallerID feed failed: Success Jul 13 15:03:34 WARNING[311316]: chan_zap.c:4777 ss_thread: CallerID returned with error on channel

HI, im having problem with callerid. Im using tdm2400P and i get this from asterisk logs -- Starting simple switch on 'Zap/4-1' [Mar 24 02:07:48] ERROR[2358]: callerid.c:564 callerid_feed: fsk_serie made mylen < 0 (-1) [Mar 24 02:07:48] WARNING[2358]: chan_zap.c:6416 ss_thread: CallerID feed failed: Success [Mar 24 02:07:48] WARNING[2358]: chan_zap.c:6516 ss_thread: CallerID

i belong to india. i am making pbx using sangoma fxo card. i want that when ever call comes to my PSTN line i should see the no from where call is coming. so i have to configures chan_dahdi.conf according to my region. i checked dahdi.conf and in that they have mentioned for india

I have 1 x100p. Caller id works fine with the beta1 release. Cvshead releases fail with a combination of checksum and ss_thread errors? I'm concerned when beta2 or the 1.2 release comes out it will not work. I have been through the configs I can't find and changes that need to be made to get CVSHEAD to work. Thanks John Hill

Hi all, I've been trying to wrap my mind around this one for several days now. How can I 'debug' the CallerID reception on a Zap/POTS channel? I have a POTS line with CallerID and a Digium TDM11B card right now. I have my signalling set to ks for both sides, can make and receive calls just fine. But I never get CallerID on incoming calls. I get the following messages: Aug 11

please help!!! I am dialing into our asterisk server(TDM400p) from the psnt. I hear our voicemail message and I press the extention 1000. The Polycom ip phone in the office rings. I pickup but neither side can hear one another. What have I done wrong? thanks sip.conf: [general] context=local-access ; Default context for incoming calls bindport=5060