[asterisk-users] FXO Hangs up automatically

Tzafrir Cohen tzafrir.cohen at xorcom.com
Tue Nov 20 12:16:57 CST 2007


On Tue, Nov 20, 2007 at 09:01:22PM +0300, Timothy Smith wrote:
> Hi,
> 
> I'm having problems using a TDM400P Card. I put my SIM card in a Nokia
> Premicell and connected it to a TDM400P card but when I make calls to
> the number, it hangs up automatically. The card also can't call out.
> Any ideas? I've searched the archives without much success. I
> appreciate all your help.
> 
> Details:
> I'm using Asterisk 1.2.17 on Fedora Core release 5 (Bordeaux). On an
> Acer Machine
> ----
> On receiving an incoming call,
> 
> Connected to Asterisk 1.2.17 currently running on pbx (pid = 5092)
> Verbosity was 16 and is now 22
>     -- Starting simple switch on 'Zap/4-1'
> Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:4502 __zt_exception: Exception
> on 16, channel 4
> Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:3687 zt_handle_event: Got
> event On hook(1) on channel 4 (index 0)

Hmmm.... it is your side (not the remote side) that can initiate an
On-Hook.

Can you try it with a simple analog phone instead?

> Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:1586 zt_disable_ec: disabled
> echo cancellation on channel 4
> Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:5683 ss_thread: waitfordigit
> returned < 0...
> Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:2493 zt_hangup: Hangup:
> channel: 4 index = 0, normal = 16, callwait = -1, thirdcall = -1
> Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:1586 zt_disable_ec: disabled
> echo cancellation on channel 4
> Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:2933 zt_setoption: Set option
> TDD MODE, value: OFF(0) on Zap/4-1
> Nov 20 20:49:32 DEBUG[6028]: chan_zap.c:1523 update_conf: Updated
> conferencing on 4, with 0 conference users
>     -- Hungup 'Zap/4-1'
> pbx*CLI>
> 
> ----
> On Trying to make an outgoing call
> 
> Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:7291 check_user_full: Setting
> NAT on RTP to 0
> Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:1415 __sip_ack: Stopping
> retransmission on '000f2300-08d000f6-4f620267-55399868 at 192.168.1.161'
> of Response 101: Match Found
> Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:7291 check_user_full: Setting
> NAT on RTP to 0
> Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:10669 handle_request_invite:
> Checking SIP call limits for device 319
> Nov 20 20:51:48 DEBUG[5110]: chan_sip.c:6267 build_route: build_route:
> Contact hop: <sip:319 at 192.168.1.161:5060>
> Nov 20 20:51:48 DEBUG[5101]: channel.c:775 channel_find_locked:
> Avoiding initial deadlock for 'SIP/319-081d8e00'
>     -- Executing Dial("SIP/319-081d8e00", "Zap/1/0004479086365389") in new stack
> Nov 20 20:51:48 DEBUG[6042]: chan_zap.c:2065 zt_call: Dialing '0004479086365389'
> Nov 20 20:51:48 DEBUG[6042]: chan_zap.c:2137 zt_call: Deferring dialing...
>     -- Called 1/0752707099
> Nov 20 20:51:49 DEBUG[6042]: chan_zap.c:4502 __zt_exception: Exception
> on 17, channel 1
> Nov 20 20:51:49 DEBUG[6042]: chan_zap.c:3687 zt_handle_event: Got
> event Hook Transition Complete(12) on channel 1 (index 0)
> Nov 20 20:51:51 DEBUG[6042]: chan_zap.c:4502 __zt_exception: Exception
> on 17, channel 1
> Nov 20 20:51:51 DEBUG[6042]: chan_zap.c:3687 zt_handle_event: Got
> event Dial Complete(9) on channel 1 (index 0)
> Nov 20 20:51:51 DEBUG[6042]: chan_zap.c:1554 zt_enable_ec: Enabled
> echo cancellation on channel 1
>     -- Zap/1-1 answered SIP/319-081d8e00
> Nov 20 20:51:51 DEBUG[5101]: channel.c:775 channel_find_locked:
> Avoiding initial deadlock for 'SIP/319-081d8e00'
>     -- Limit Data for this call:
>     -- - timelimit     = 0
>     -- - play_warning  = 0
>     -- - warning_sound = (null)
> Nov 20 20:51:51 DEBUG[5110]: chan_sip.c:1415 __sip_ack: Stopping
> retransmission on '000f2300-08d000f6-4f620267-55399868 at 192.168.1.161'
> of Response 102: Match Found
> Nov 20 20:51:51 DEBUG[6042]: chan_sip.c:3051 sip_rtp_read: Oooh,
> format changed to 256

So what exactly is wrong here? SIP talking to Zap with a ulaw format.
Anything wrong?

> 
> 
> The Call doesn't go through
> ---
> Out put of `lspci`
> .
> .
> 00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
> Modem/ISDN interface
> .
> .
> .
> ---
> Output of `lsmod`
> Module                  Size  Used by
> wctdm                  37184  4
> .
> .
> .
> -----
> Output of /proc/zaptel/1
> 
> [root at pbx ~]# cat /proc/zaptel/1
> Span 1: WCTDM/0 "Wildcard TDM400P REV H Board 1"
> 
>            1 WCTDM/0/0 FXSKS (In use)
>            2 WCTDM/0/1 FXOKS (In use)
>            3 WCTDM/0/2 FXOKS (In use)
>            4 WCTDM/0/3 FXOKS (In use)
> [root at pbx ~]#
> 
> ----
> Output of ztcfg -vvvv
> 
> [root at pbx ~]# cat /proc/zaptel/1
> Span 1: WCTDM/0 "Wildcard TDM400P REV H Board 1"
> 
>            1 WCTDM/0/0 FXSKS (In use)
>            2 WCTDM/0/1 FXOKS (In use)
>            3 WCTDM/0/2 FXOKS (In use)
>            4 WCTDM/0/3 FXOKS (In use)
> 
> ----------
> [root at pbx ~]# ztcfg -vvvv
> 
> Zaptel Configuration
> ======================
> 
> 
> Channel map:
> 
> Channel 01: FXS Kewlstart (Default) (Slaves: 01)
> Channel 02: FXO Kewlstart (Default) (Slaves: 02)
> Channel 03: FXO Kewlstart (Default) (Slaves: 03)
> Channel 04: FXO Kewlstart (Default) (Slaves: 04)
> 
> 4 channels configured.
> 
> [root at pbx ~]#
> 
> ----------------
> My /etc/zaptel.conf
> 
> [root at pbx ~]# cat /etc/zaptel.conf
> fxsks=1
> fxoks=2-4
> loadzone = us
> defaultzone=us
> [root at pbx ~]#
> 
> ------------------
> My /etc/asterisk/zapata.conf
> 
> [root at pbx ~]# cat /etc/asterisk/zapata.conf
> [channels]
> group=2
> signalling=fxo_ks
> context=outgoing
> callerid="Extensions"
> channel => 2-4
> 
> group=3
> signalling=fxs_ks
> context=analog-incoming
> channel => 1
> [root at pbx ~]#
> 
> --------
> Out put of zap show
> pbx*CLI> zap show channels
>    Chan Extension  Context         Language   MusicOnHold
>  pseudo            from-pstn-celte
>       1            from-pstn-celte
>       2            outgoing
>       3            outgoing
>       4            outgoing
> switch1*CLI> zap show status
> Description                              Alarms     IRQ        bpviol     CRC4
> Wildcard TDM400P REV H Board 1           OK         0          0          0
> pbx*CLI>
> 
> ------------
> Extract of /etc/asterisk/extensions.conf
> 
> [analog-incoming]
> exten => s,1,Answer
> exten => s,2,Dial(SIP/307)
> exten => s,3,Hangup
> ------
> 
> Thank you very much for your assistnce.
> 
> Warm Regards,
> 
> _______________________________________________
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-- 
               Tzafrir Cohen
icq#16849755              jabber:tzafrir.cohen at xorcom.com
+972-50-7952406           mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com  iax:guest at local.xorcom.com/tzafrir



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