similar to: Asterisk, POTS and plain handsets

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

Hello, I'm having a problem in voicemail check attempts from SIP-based phones. I've searched a ton of docs but don't see anyone else having a similar issue. I have a TDM22B with two non-sip phones connected to it as well as several SIP phones including a GXP-2000 and some X-Lites. Users of the real phones in the same context can pick up and dial *8 to get to VoiceMailMain() just

I now have an OpenVox A400P and it is working well. Thanks to Ade Vickers for the recommendation, which I second. However, I need to make a slow transition between a conventional multiple-extension setup and a full VoIP network on these premises. So at the moment the Asterisk box shares the PSTN connection with several conventional analogue handsets. The desired result for an incoming call is

ERROR CONDITION --------------- -- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack -- Called 2000 -- SIP/2000-0ead is ringing -- SIP/2000-0ead answered SIP/2001-f6c4 -- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead Have searched web and archive w/o good results. Thks in advance for any help, Dave sip.conf -------- [general] port =

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

Of course I am not a kernel expert, so .. please be patient. I am investigating on my zaptel/zapata problem. As the main error message asterisk quits on mentions <'/dev/zap/channel': No such file or directory> I went peeking over there. [Asterisk Verbose Error Mar 13 20:43:35 WARNING[5779]: chan_zap.c:763 zt_open: Unable to open '/ dev/zap/channel': No such file or

Asterisk 11.1 We have a situation where one of our incomings POTS lines will not answer. There are 2 lines configured by the Telco as a rollover group (rings the line that is not busy) and they feed into a Digium AEX410 on the server. The most recent time this happened, I did a /etc/init.d/dahdi status and saw this: ### Span 4: WCTDM/1 "Wildcard AEX410" *53 FXO FXSKS

No Caller ID comes from the FXO line ( The caller id is on and is working with a standard phone) in zapata.conf everything looks fine usecallerid=yes hidecallerid=no When the call comes in there are some warnings in Asterisk Console -- Starting simple switch on 'Zap/4-1' Jun 22 11:20:24 NOTICE[213006]: callerid.c:281 callerid_feed: Unknown IE 17 Jun 22 11:20:24 NOTICE[213006]:

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

I have a normal backup phone (and an alarm panel) sharing the POTS line with the Digium X101P FXO: | | Wall |>---+------X101P FXO as Zap/5 | | | Phone & Alarm Whenever the Phone is used, Asterisk sees a 'false ring' signal immediately when the phone is hung up. The Alarm panel dials out nightly at around 1AM, and each time it completes the call, Asterisk

Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: app_queue.c:3939 queue_exec: unable to join queue "myqueue" In extension file: Queue(myqueue|t|||120) And my agents are joining in following

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, again. I've spent a week trying to get asterisk to work on FreeBSD unix, with some success. Everything works until I plug the box into the TELCO line and then the line goes off-hook and stays that way. So I bit the bullet and decided to install the application on a fresh linux install. Not to start an OS war, here, but linux is ... difficult ... for an old unix hand to get his mind

Hi, I'm trying to get up and running a TDM400 with a standard italian pots line but i'm having problems at getting asterisk to detect when the line get answered on outgoing calls. I'm using asterisk 1.6 beta 9 with zaptel 1.4.11. I tried with and without answeronpolarityswitch=yes but it didn't change anything at all. With callprogress=yes answer get never detected. With

Quick and dirty question: for the TDM410P I must use the wctdm24xxp driver? Att Vin?cius Fontes Desenvolvimento Canall Tecnologia em Comunica??es Ltda.

Hi- I'm having a recurring problem with Windows XP and a Samba share on logon. More specifically, I have a Samba (v 3.0.24 on Debian Etch) share with share permissions (not a domain) mapped to a drive in Windows XP Pro. Whenever someone logs onto an XP Pro computer in the network the drive isn't there. Here's an example: Say there's a document on the shared drive and a

Hi, Before I report a bug on http://bugs.digium.com, I would like to know if someone is seeing the same error message. Personally I am not using wctdm24xxp but other modules such as wcte12xp and wctdm. The latter modules load fine and are compiled with pci_register_driver as expected. The only module that seems to require the deprecated function pci_module_init is wctdm24xxp. Is this normal?

I am not sure what the problems are and the reasons for the basic 64K modems used in VOIP are. I understand the compressed codecs that get the bandwidth down to 20-30 K. And perhaps the 64K units give much better potential audio than you would get on a normal POTS line. But, as phone circuits VOIP/SIP doesn't seem to perform as well as plane old phones. Multiple transcodings cause issues.

i have 6 pots lines coming in from the outside world (but we are reducing to 4) all the lines have the same phone number. i have 40 analog telephones that need to be connected to them. one way (i think) i could do this is to have a asterisk box with a tdm400p with 4 fxo's connected to the pots, also in the asterisk box would be a pair of t100p each connected to a tsu-600, the tsu-600 would