similar to: False hangups with TDM400P and Kewlstart

Hi Most of the threads in the list archive relating to X100P and hangups are about not detecting hangups. We have got the opposite problem. We have experienced an increased number of false hangups when connecting an X100P to an analog port of an ISDN terminal adapter. It happens more frequently on incoming calls than it does on outgoing calls. Often hangups occur after about 3-4 minutes into the

I just purchased a T100p from digium and a Carrier Access Access Bank 1 channel bank (12fxs/12fxo). I have the setup partially working thanks to some help from IRC. However I still have the following issues I can't seem to resolve 1. When calling into the system from the PSTN call hangup is not detected. * leaves line in use until it is shutdown. 2. When calling an analog phone connected to

Hello, We are using a TDM400p with 4 FXOs and SIP phones in a high call-volume environment. At least twice a day there are complaints of 'dropped calls'. Examining the debug logs, I see that in each case, an "on hook" event is detected, followed by the zap channel being hung-up and * saying "BYE" to the sip phone: Jun 23 14:17:22 DEBUG[2441232]: Exception on

Hello, I am having an issue with hangups being handled within Asterisk. Right now, when an inbound call hits the Asterisk box, Asterisk picks up the call just fine. When the caller enters an extension to call, the Asterisk dials out on Zap/3 and rings the extension with no problem. If the extension is answered, there is no problem. If the caller hang's up before the phone is answered,

Hello! We are using the Digium 405PP card, and getting the following messages: Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 My config file is below. We are trying to set up D-Channel on channel 24, 1-23 in trunk group 1,

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

Hello, is there a full guide to what kewlstart is supposed to do with FXO or FXS lines ? is it only applicable to one of the interfaces FXO -or- FXS but not both ? I asked earlier if FXS lines can be made to reverse polarity, and someone else pointed out that the chipset on the FXS ports seems to support it, perhaps the driver in the asterisk zaptel interface module needs to be modified to support

Hi All, Which is the best signalling to use when connecting an FXS inteface on a TDM400 to a standard telephone. I see that all examples use fxo_ks, but it is my understanding that kewl start is really designed for connections to the CO so that hangup etc. can be detected. So does it make any sense to configure a telephone for fxo_ks? Or should it be configured for fxo_ls? Regards Garry Taylor

Hi, Try as I might, I can't get hangups detected on a Zap channel with loop start lines. So, after someone leaves a voicemail and then hangs up, Asterisk doesn't know it, exits VoicemailMain2, and loops back to the corporate greeting, tying up the line even though the outside caller has hung up. Therefore, I've added the following hideous hack - er, code - to voicemail2.c. It

Specs FC2, Asterisk 1.0.0, Zaptel 1.0.0 TDM400P Port 1 FXS Port 4 FXO Standard analogue handset plugged in with pstn line. Problem: When I go to dialout it drops numbers on the outgoing number. Keys dialed from handset were 9 0418800185 I tried hitting the keys slowly as well as at my normal speed, all tones are heard in the handset for all numbers.

All, I'm having trouble getting the X100P working. Lsmod shows : zaptel 179808 0 I did a . # modprobe zaptel and here is my zaptel.conf (comments omitted) __SNIP__ fxsks=1 loadzone = us defaultzone=us __SNIP__ Here is zapata.conf __SNIP__ [trunkgroups] [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 usecallerid=yes hidecallerid=no

Hello all. I have Asterisk connected to an Adit 600 channel bank with a TE110P and the channel bank is connected to a PBX providing dialtone to the PBX with fxo_ks signalling. When a call between the PBX and Asterisk completes there is a momentary battery drop/reversal or something that signals the PBX that the other side has hung up and then the PBX hangs up. This all works fine. However,

Sorry to post such a newb set of questions but I have been hammering about trying to get Asterisk running on FC2 machine reading everything available (I think that is what stuffed me, shouldn't have read it all :-) ). Config FC2 running Asterisk 1.0.0, with the h323 compiled in and installed correctly. Amazingly enough I have everything compiled correctly and installed. I am running a

Scenario, I got some very good help earlier from Joseph getting me up and started but I have a couple of small problems still. Setup: FC2 Asterisk 1.0 Zaptel 1.0, TMP400P, FXS Port 1 FXO Port 4 Analog dialout line and Analog handset plugged in. Problems: 1. Incoming calls work and the phone rings and can be answered no problems, (although I wouldn't mind being able to adjust the ring but

I have an asterisk server running on Redhat 8.0 with a Digium TDM400P w/4 FXO modules (TDM04P) There are 2 lines going into the Digium card. One line is a Vonage digital line, and the other line is a Comcast voice line. I have a SIP Grandstream 100 phone connected to the Asterisk server. I also have IAX configured with FWD. The problem is that on occasionally, after talking for about 20

Steve, Here is the config, I pulled from my server, that works with D'Link Phones: Main Menu -------------------------------------------------------------------------------- SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) ;bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) bindaddr = 67.109.153.236 disallow=all ;allow=ilbc allow=gsm allow=ulaw

Hi, I had ordered a TDM40B and developers kit a few months ago. I have everything installed and working, with one exception - sound quality. When placing a call it sounds like a very bad cordless phone - lots of hiss / static in the background. This even happens with the dialtone, though it is much worse one the call is connected. This does not occur when the phone is directly connected to the

Hi i have buy a Digium TDM400P with 1 fxo modules TDM01B for connect my asterisk to a french analog line. In my zaptel.conf, i have: loadzone=fr defaultzone=fr fxols=3 when i load the module, i have in logs: Nov 24 06:13:40 gw kernel: Freed a Wildcard Nov 24 06:13:40 gw kernel: Zapata Telephony Interface Unloaded Nov 24 06:13:40 gw zaptel: Removing zaptel module: succeeded Nov 24

I was really looking forward to Asterisk 1.0 et al, but it is a major disappointment. I have never experienced any Asterisk release that was interacting with Digium hardware so unreliably. Asterisk hangs up on every outgoing PSTN call (via Zaptel) as soon as the call is being picked up at the other end. I have tried various X100P (original Digium) cards, various phone lines and just about every

I have asterisk FXO module on a TDM400P hooked to an Intertel Single Line Card. I can place Intertel intercom calls to Asterisk (both SIP and analog phones) and the reverse, but transfering calls doesn't work. Here what the transfer looks like: Intertel FXS > Transfer > Asterisk FXO > Asterisk FXS > any extension When the call is answered Asterisk hangs up. Intertel