similar to: false answer on zaptel

Hello! I also post here this messages, maybe it's a bug. I have a problem with Cygwin OpenSSH, I hope somebody can help me out. Since we reinstalled our machine we can't connect any external hosts, but we can connect the gateway server. The same box is when booted up with linux (Debian unstable) just works fine. The problem is, that ssh stops at "expecting

I am looking for someone that could share their code for this function: Outgoing call -> macro that checks if line is (not human) or machine, fax, busy, subscriber problem and other fault tones -> if human connect to agent else hangup and write status to cdr. Need help with this! Regards / Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I am working on an outdial project and the Asterisk box is connected behind a PBX via SIP. When a call from the Asterisk box is routed out over the PRI attached to the PBX I am not getting proper call progress. The PBX is indicating that the call is answered while it is still ringing at the far end. Does anyone have any suggestions on how I should go about waiting for a variable number

Hi all.. I have a problem with my asterisk installation. I'm using a Wilcard X100P clone in Spain. Incoming calls work fine, but when I make a outgoing call, a hear the ringing, and the peer phone ring, when the peer answer, asterisk hangup the call, or say busy. This is my conf: zaptel.conf: --------- loadzone = es defaultzone=es fxsks=1 zapata.conf ---------- [channels]

I've just had polarity reversal provisioned by our telco to test hangup detect with a TDM400P I've set hanguponpolarityswitch=yes in zapata.conf When I start Asterisk I get "ignoring hanguponpolarityswitch" in /var/log/asterisk/messages I assume that the option is either not valid or conflicts with another setting somewhere. Any ideas?

> From: "Patrick Healy" <pjhealy@healyville.com> > Subject: [Asterisk-Users] Using call.sample on Zap hardware - > Answering problem > I've got a X100P connected to a POTS line and am using it to call out to > play a recorded message. I drop a copy of sample.call into > /var/spool/asterisk/outgoing and Asterisk picks up the line and initiates > the call.

There are a few mentions in the wiki [1] about a zapata.conf flag "hanguponpolarityswitch". It is meant to cause Asterisk to detect a hangup when the line polarity switches at the end of the call. The wiki mentions using the flag in zapata.conf but when I do Asterisk "ignores" it: Apr 12 17:59:38 WARNING[12804]: chan_zap.c:10875 setup_zap: Ignoring hanguponpolarityswitch

---------- Forwarded message ---------- From: Arun Kumar Chaudhary <uniquearun04 at gmail.com> Date: Sat, Jun 21, 2008 at 4:51 PM Subject: Detection of Answer, hangup,busy etc while using Dial command To: asterisk-users at lists.digium.com. Hi Guys, I am in kanpur, India. I am using Dial() command in my phpagi script. I am unable to detect whether it is connected to the dialed number, if

hi: In my zapata.conf i have 4 fxo configured channels,for fxo number 1 to 3 i added polarity reversal property but for fxo number 4 i didnt add polarity reversal property but it still giving me on cosole that fxo number 4 is polarized (because the line on fxo number 4 is not polarized). what i want to do is to not let polarity reversal take effect on fxo number 4. that what i have in my

Afternoon, I was hoping someone could point me in the right direction. I have an asterisk PBX deployed in China using a TDM400P based card. The incoming calls are being picked up correctly, but are not being hung up. I suspect that this might be an issue with the signaling that has been selected. If anyone here has deployed asterisk in china using an analog card, it would be a great help

Dear list, We have an weird problem with our FXO card (TDM01B). When I made a call using this channel, all goes well, the called phone rings but when the called phone answers the call. In me handset I can hear an weird sound like a "Clack". I tryed diferents TDM cards and modules, and my zapata.conf is like, language=en context=from-zaptel switchtype=national usecallerid=yes

Hi all, I've just got an OpenVox A400P card with 1 FXO and 1 FXS module and I am just trying to get it working. But no luck as of yet. In /etc/zaptel.conf I've set the following options: fxsks=2 fxoks=1 loadzone=se defaultzone=se And in /etc/asterisk/zapata.conf I've not sure what to set exactly. For example, under [trunkgroups] what to specify there? Under [channels I set something

I have a very strange problem I'm hoping someone has encountered already. I've scoured the internet for an answer to this one. My phone company provides no disconnect supervision. Hence I'm forced to use the busydetect feature. I have a TDM400 with two FXO ports. If I call from an internal extension to a landline and then hangup the landline Asterisk detects the busy signal

Hi With an FXO module + Zaptel, I'd like to know if there are ways to know when the remote party has answered the phone, whether calling through a callfile or by sending DTMF's. I read about {CHANNEL(state), ChanIsAvail(), and ${DIALSTATUS}: Are those reliable ways to know when the channel is available for dialing out and the call has been answered?

static void senddialevent(struct ast_channel *src, struct ast_channel *dst) { manager_event(EVENT_FLAG_CALL, "Dial", "Source: %s\r\n" "Destination: %s\r\n" "CallerID: %s\r\n" "CallerIDName: %s\r\n" "SrcUniqueID: %s\r\n" "DestUniqueID: %s\r\n" "CDRUserfield: %s\r\n", src->name,

Hi, Im working with the following versions: -asterisk-1.2.12.1 -zaptel-1.2.9.1 And with the following card: 00:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 8085:0003 Flags: bus master, medium devsel, latency 32, IRQ 201 I/O ports at c800 [size=256] Memory at fe000000 (32-bit, non-prefetchable)

I've got an old analog PBX and I'm trying to connect an FXO port on my asterisk server to one of the extensions on the old PBX. This should work as en extension on the old PBX should be providing dialtone, battery current and ring voltage. However, when the old PBX "hangs up" asterisk doesn't appear to be detecting the hangup (the DAHDI channel stays in use). Can anyone

Hello All, I'm trying to connect Asterisk to an Executone phone system with an analog DID card and I'm hoping someone can help me figure out what I'm doing wrong. The Executone DID card provides battery to the telco, when the telco wishes to dial a DID it goes off-hook, waits for a wink from the Executone and then dials the last three digits on the number with pulse (as opposed

Hello, my config : aah 2.6 (asterisk 1.2.4) , centos 4.2, 1 TDM11B (1 Fxo / 1 fxs ), 1 phone, 1 softphone I'm in France When someone from PSTN calls and hangs up before the call is answered, internal extension keeps ringing until timeout occurs. PSTN line keeps busy. Hangup detection doesn't work. I've played with different paremeters (callprogress, busydetect, busycount,

Hi, I'm having trouble with dialing out on analog lines. Asterisk can't seem to detect "answers". I have two zap groups. Group 1 is connected to an external analog PSTN provider. This group seems to work fine, especially with "answeronpolarityswitch". Group 2 is a group of "GSM gateways", ie. devices that host SIM cards so that you can dial from any PBX