I’m running Asterisk 1.2 and have setup a TDM800P card.
I’m facing a problem with OG calls.
I’ve configured to call a zap channel on an extension.
Problem is…
The phone connected to the FXS zap channel is not ringing. But when I go offhook (pick up the phone) there is 2 way voice loud and clear.
I saw the status of the channel it says OnHook (see below)
[i]
– Executing Dial(“SIP/1230-093e9758”, “Zap/1/1250”) in new stack
– Called 1/1250
– Zap/1-1 answered SIP/1230-093e9758
localhost*CLI> zap show channel 1
Channel: 1CLI>
File Descriptor: 12
Span: 1
Extension:
Dialing: no
Context: mobile-land
Caller ID:
Calling TON: 0
Caller ID name:
Destroy: 0
InAlarm: 0
Signalling Type: FXO Loopstart
Radio: 1
Owner: Zap/1-1
Real: Zap/1-1
Callwait:
Threeway:
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: yes
Relax DTMF: yes
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps unless TDM bridged, currently OFF
Actual Confinfo: Num/0, Mode/0x0000
ctual Confmute: No
Hookstate (FXS only): Onhook
[/i]
This is my extensions.conf ( I dial 1250 from my SIP phone(xlite) )
exten => 1250,n,Dial(Zap/1/1250)
thanks for the replies.
Vinod
Hi Loan,
thanks for the reply, and sorry that I should hve mentioned what all i’ve tried already. inline for your suggestions…
-
Zap/1/1250 <- here you could use only Zap/1 as there is no “phone number” to be dialed (this is used only on FXO ports)
…Yes, thats what I tried first, I added the phone number later. but still no luck.
-
Maybe the rings generated by TDM card is not detected by your analog phone.
… in this case i assume the “zap show channel 1”, must indicate Ringing status. whereas it is showing Onhook. (pls correct if i’m wrong here)
a. have you tried with another phone?
… yes I have. infact i unplugged the phone used for the office pbx and tried with this.
b. maybe you could play with “distinctive ring styles” - more details here -> voip-info.org/wiki/view/Aste … P+channels
… I tried adding a distinctive alert, as “exten => s,1,Dial(Zap/1r2)” … still the same.
apart from that…
I have immediate=no in zapata.conf
I am thinking this might be some configuration issue, since the channel status itself is not changed. But I dont think extensions.conf would need anything more to make a phone ring (but then, that’s what i’m not sure of)
yeah … the power connector is there.
and also, when I make a call and go Offhook, I am able to hear the voice from the callee.
If I go offhook when No call is present, there is an echo running which shorts my voice to the earpiece, so atleast the power may not be the problem. (no access to oscilloscope or spare resistors here… so this is how I checked)
just fyi: there has been a similar discussion but, without result… over here. bit.ly/9PMTmT
thanks
Vinod
well - something is wrong as you have to hear a dial tone when you pick up your analog phone - or I am wrong?
have you tried to change the signalling from fxo_ls to fxo_ks (in zaptel.conf and zapata.conf)?
please set verbose and debug level to 10 and post a log caption (/var/log/asterisk/full) when an incoming call is sent to the analog extension. maybe it will help us to see if an error is reported or not.
HTH,
Ioan
Yeah … the dial tone is supposed to be there. but it is not.
in zapata.conf, the signaling is fxo_rx . when I change it to fxo_ks chan_zap.so load error in encountered.
This is the verbose output of making a call from SIP phone and trying to ring a zap channel on that from "full"
May 4 14:15:27 VERBOSE[2800] logger.c: – Remote UNIX connection
May 4 14:15:37 DEBUG[2814] chan_sip.c: Setting NAT on RTP to 0
May 4 14:15:37 DEBUG[2814] chan_sip.c: Checking SIP call limits for device 1230
May 4 14:15:37 DEBUG[2814] chan_sip.c: build_route: Contact hop: sip:1230@10.232.44.94:38296
May 4 14:15:37 VERBOSE[2829] logger.c: – Executing Dial(“SIP/1230-09fccc70”, “Zap/1/1250”) in new stack
May 4 14:15:37 VERBOSE[2829] logger.c: – Called 1/1250
May 4 14:15:37 VERBOSE[2829] logger.c: – Zap/1-1 answered SIP/1230-09fccc70
May 4 14:15:37 DEBUG[2814] chan_sip.c: Stopping retransmission on ‘c45b5a43f768054dMDdiZTNkODRhMWM3ZjVmZGEwMGI2ZTc3YzA2YTY2N2M.’ of Response 1: Match Found
May 4 14:15:43 DEBUG[2829] chan_zap.c: Got event Ring/Answered(2) on channel 1 (index 0)
May 4 14:15:43 DEBUG[2829] channel.c: Got a FRAME_CONTROL (12) frame on channel Zap/1-1
May 4 14:15:43 DEBUG[2829] channel.c: Bridge stops bridging channels SIP/1230-09fccc70 and Zap/1-1
May 4 14:15:44 DEBUG[2829] chan_zap.c: Got event On hook(1) on channel 1 (index 0)
May 4 14:15:44 DEBUG[2829] channel.c: Got a FRAME_CONTROL (13) frame on channel Zap/1-1
May 4 14:15:44 DEBUG[2829] channel.c: Bridge stops bridging channels SIP/1230-09fccc70 and Zap/1-1
May 4 14:15:47 DEBUG[2829] channel.c: Didn’t get a frame from channel: SIP/1230-09fccc70
May 4 14:15:47 DEBUG[2829] channel.c: Bridge stops bridging channels SIP/1230-09fccc70 and Zap/1-1
May 4 14:15:47 DEBUG[2829] chan_zap.c: Hangup: channel: 1 index = 0, normal = 15, callwait = -1, thirdcall = -1
May 4 14:15:47 DEBUG[2829] chan_zap.c: disabled echo cancellation on channel 1
May 4 14:15:47 DEBUG[2829] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1
May 4 14:15:47 DEBUG[2829] chan_zap.c: Updated conferencing on 1, with 0 conference users
May 4 14:15:47 VERBOSE[2829] logger.c: – Hungup 'Zap/1-1’
May 4 14:15:47 DEBUG[2829] app_dial.c: Exiting with DIALSTATUS=ANSWER.
May 4 14:15:47 VERBOSE[2829] logger.c: == Spawn extension (default, 1250, 2) exited non-zero on 'SIP/1230-09fccc70’
May 4 14:15:47 DEBUG[2829] chan_sip.c: update_call_counter(1230) - decrement call limit counter
May 4 14:15:47 DEBUG[2829] pbx.c: Function result is '“1230” <1230>'
May 4 14:15:47 DEBUG[2829] pbx.c: Function result is '1230’
May 4 14:15:47 DEBUG[2829] pbx.c: Function result is '1250’
May 4 14:15:47 DEBUG[2829] pbx.c: Function result is 'default’
May 4 14:15:47 DEBUG[2829] pbx.c: Function result is 'SIP/1230-09fccc70’
May 4 14:15:47 DEBUG[2829] pbx.c: Function result is 'Zap/1-1’
May 4 14:15:47 DEBUG[2829] pbx.c: Function result is 'Dial’
May 4 14:15:47 DEBUG[2829] pbx.c: Function result is 'Zap/1/1250’
May 4 14:15:47 DEBUG[2829] pbx.c: Function result is '2010-05-04 14:15:37’
May 4 14:15:47 DEBUG[2829] pbx.c: Function result is '2010-05-04 14:15:37’
May 4 14:15:47 DEBUG[2829] pbx.c: Function result is '2010-05-04 14:15:47’
May 4 14:15:47 DEBUG[2829] pbx.c: Function result is '10’
May 4 14:15:47 DEBUG[2829] pbx.c: Function result is '10’
May 4 14:15:47 DEBUG[2829] pbx.c: Function result is 'ANSWERED’
May 4 14:15:47 DEBUG[2829] pbx.c: Function result is 'DOCUMENTATION’
May 4 14:15:47 DEBUG[2829] pbx.c: Function result is '(null)'
May 4 14:15:47 DEBUG[2829] pbx.c: Function result is '1272962737.2’
May 4 14:15:47 DEBUG[2829] pbx.c: Function result is ‘(null)’
hey - there is no fxo_rx signalling in analog channel (ZAP or DAHDI).
and if you do not change to fxo_ks in both file (/etc/zaptel.conf and /etc/asterisk/zapata.conf) you have errors.
in case you do not manage to modify the signalling to fxo_ks (from fxo_ls) please post both files here and we’ll try to help you.
BR,
Ioan.
oh Gosh!!! what a blunder…
I replace fxo_rx with fxo_ls and it rings!!
Thanks a lot Loan for staying with me and helping me solve the problem.
although apparently … fxo_rx is a valid signalling (checked the chan_zap.c, it does the follwing
chan_conf.signalling = SIG_FXOLS;
chan_conf.radio = 1; )
And so, it marked the zap channel as answered without ringing the phone. and therefore when I go offhook I’m able to hear the voice.
Now dont ask me why rx came in my file the first place… have no idea. 
regards,
Vinod
aha - good to know - have fun with your Asterisk server 