search for: zap

Every so often, and it seems that it happens only when a call is in progress, all 24 Zap channels get reset. All channels are opened and then timeout. This causes the in-progress calls to terminate. There are no corresponding Red/Yellow alarms on wither the PBX or Asterisk although we do receive a fair amount of Blue Alarms. The Asterisk server is connected to a legacy PBX through a...

...uting DISA seems to prevent any DTMF detection capability when using the FXO port of the TDM400. Below, config A and B and their debug logs. In Config A I use Authenticate() instead of using DISA password since it demonstrates that it's DISA that seems to prevent DTMF detection when using Zap/1. Otherwise DISA works flawlessly when calls are coming from FXS port (TDM400), IAX, SIP channels.... and we have absolutely not other problem detecting DTMF that we are aware of... I see no active bug related to DISA at bugs.digium.com... Any idea? Ben. *Code:* --------------------------...

...! Got reject for frame 65, but we only have others! May 15 20:27:49 WARNING[4033]: PRI: !! Got reject for frame 106, but we only have others! May 15 20:30:13 WARNING[4033]: PRI: !! Got reject for frame 114, but we only have others! May 15 20:36:10 WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! May 15 20:37:48 WARNING[4033]: Avoided initial deadlock for 'Zap/64-1', 10 retries! May 15 20:38:37 WARNING[4033]: Failed to write frame May 15 20:39:08 WARNING[4033]: Avoided initial deadlock for 'Zap/64-1', 10 retries! May 15 20:39:14 WARNING[4033]: Avoid...

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,Di...

...service. Here's a typical scenario: 1) Call 765-1574, 2) When they answer, press flash and call 735-1056, 3) Press Flash again and I have a three way call 4) Try calling 820-2726 on the other line and I get fast busy. Asterisk indicates that it was "Unable to create channel of type 'ZAP'" when I attempt the threeway call. As a result, the second call (735-1056) is placed on the other ZAP channel so there's none left when I attempt the third call (820-2726) on the other FXS line. There's no problem with my Bell service as I can do three way calls on BOTH my F...

...transfer led to a crash of asterisk, without leaving a core or any utterances in messages or debug file. It looks like the zombie which was created during the MASQ-transfer was not cleaned up... But why did it start a Dial??? And... why does Asterisk die when this happens?? Thanks!!! Michiel -- Zap/32-1 answered Zap/6-1 -- Stopped music on hold on Zap/6-1 -- Starting simple switch on 'Zap/32-2' -- Started three way call on channel 32 -- Started music on hold, class 'default', on Zap/6-1 -- Executing Macro("Zap/32-2", "stdexten2|6112|Zap/34|20...

Hi, i have an asterisk box running with E100P (E1) line as PSTN gw. Sometimes zap channels hang and i couldn't make any PSTN calls but SIP calls are still fine. When this happens I also couldn't restart/reload asterisk from the CLI. I have to kill the asterisk process and run safe_asterisk again. any ideas? asterisk*CLI> show channels Channe...

Hi I have 3FXO trunks called ZAP-25,ZAP-26 and ZAP-28 and T1 Channnel bank I get this deadlock problem when 2 incoming call from FXO(Here ZAP-28 and then ZAP-26) wants to dial same channel (Here ZAP-1). In this senario ZAP-1 first answer ZAP-28 and thne ZAP-26 wants to call ZAP-1 but it time out and goto voicemail after that Z...

I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs line coming into a Digium TDM01B. It appears to not be getting CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound and outbound calls work fine but there is just no CID on inbound for this channel.The incoming route for the channel is Za...

I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used line. Would the following work for 911 calls? [e911] exten => 911,1,ChanIsAvail(Zap/1) exten => 911,2,Dial(Zap/1/911) exten => 911,3,Hangup() exten => 911,102,ChanIsAvail(Zap/4) exten => 911,103,Dial(Zap/4/911) exten => 911,104,Han...

...tem setup witht he following configuration Telco <-> Asterisk E405P <-> BP250 The system seem to work perfectly on 1.0.9 for a very long time but there is some functionality we wanted to take advantage of in the 1.2 version branch so we upgraded. Currently running Asterisk 1.2.4 Zaptel 1.2.3 (noticed 1.2.4 is out will upgrade next weekend) Libpri 1.2.2 The problem we are getting is wierd but :- Sorry about the timings looking wierd but you have to allow a fudge factor of anywhere upto 12 hours when dealing with reports from on-site personel. * Wednesday ~ 9.30am All calls...

Hello to all: When I make a call of extension (FXS) to extension (FXS) (TDM40B) status of the call is the following one: #### Case call answered --------------------------------------------------- Log init -- Starting simple switch on 'Zap/4-1' -- Executing Macro("Zap/4-1", "stdexten|103|Zap/2") in new stack -- Executing Dial("Zap/4-1", "Zap/2|30|Ttr") in new stack -- Called 2 -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/4-1 -- Attempting n...

...erisk, I have enabled DTMF CID, the data is arriving to the asterisk but asterisk isn't interpretating it: its my full log: 1. Mar 10 16:26:03] DEBUG[8715] dsp.c: dsp busy pattern set to 0,0 2. [Mar 10 16:26:03] VERBOSE[9274] logger.c: -- Starting simple switch on 'Zap/4-1' 3. [Mar 10 16:26:03] VERBOSE[9274] logger.c: -- Executing [s at from-trunk:1] Set("Zap/4-1", "__FROM_DID=s") in new stack 4. [Mar 10 16:26:03] VERBOSE[9274] logger.c: -- Executing [s at from-trunk:2] Wait("Zap/4-1", "4&quot...

...ave followed the instructions to the best of my knowledge. When a fax comes in, the system seems to detect OK but does ot manage to make the fax to pdf to email leap. Here is what I saw in the CLI when I tested. Any help would be appreciated. Thanks! Wiley -- Starting simple switch on 'Zap/2-1' -- Executing GotoIf("Zap/2-1", "1?from-pstn-reghours|s|1:") in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf("Zap/2-1", "0?from-pstn-reghours-nofax|s|1:2") in new stack -- Goto (from-pstn-reghours,s,2) -- Executin...

Calling out on PSTN over a TDM400P seems to generate a Red Alarm - whatever that is. I have another extension on the PSTN, and I can dial out over that. zttool shows no alarms. asterisk*CLI> zap show status Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV I Board 1 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) -- Starting simple switch on 'Zap/2-1' -- Executing [2560597 at...

On 1.2.x I have a random problem when a Zap/x channel flashes to transfer or make a three way call. The Zap/x-2 channel is created and the transfer or three way proceeds, but on hangup the Zap/x-1 channel fails to destroy the old bridge and asterisk goes crazy logging the problem. Here is an example debug log. This happens only once a...

...e TDM-04B is supposed to answer, play some sounds, and hang up. The TDM-04B detects the ring, but doesn't actually answer -- my cell phone never stops ringing, and the call plan loops the following information even after I've hung up the cell phone: -- Starting simple switch on 'Zap/1-1' -- Executing Answer("Zap/1-1", "") in new stack -- Executing DigitTimeout("Zap/1-1", "5") in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout("Zap/1-1", "10") in new stack -- Set Response Timeou...

Skipped content of type multipart/alternative-------------- next part -------------- asterisk1*CLI> soft hangup Zap/1-1 Requested Hangup on channel 'Zap/1-1' == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Zap/1-1' in macro 'exten-vm' == Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple sw...

...up exten => s,i,Hangup exten => s,h,Hangup I know I must be missing something simple, but here is the output from dialing my home, answering the call, making another call to my home in order to do a callwaiting transfer, doing a # and a *43016 (3016 is the sip number I'm answering the Zap channel on). I am using an analog phone off of a Cisco ATA186 for extension 3016. I replaced the number I was dialing from into my home with 5555551212. astera*CLI> -- Starting simple switch on 'Zap/1-1' -- Executing Wait("Zap/1-1", "1") in new stack...

Ok, I've seen discussion before on doing transfers (attended and unattended), there seems to be much confusion over it. As things sit, I've been trying (unsuccessfully) to do transfers with a zap channel (analog phone attached to TDM400). I have no idea what I'm missing. My current understanding is that I need to have transfer=yes in zapata.conf, do a flash hook, dial the 2nd number, flash hook again and we're linked (attended). Then if I hang up the call will be transfered. Ho...