Problem dialing out through ZAP

[daniel..]

I have a problem with my PIAF server:

When dialing out through a ZAP channel (Digium X400), the call hangs up immediately when the recipient gets the phone. Incoming calls work fine. Any ideas why this is happening?

My status page is the following:

Status Version 1.2.9 released on Date 042310
********************************************************************
*            PBX in a Flash Version  Daemon Status                 *
*                      Running Asterisk 1.4                        *
********************************************************************
* Asterisk  * ONLINE  * Zaptel    * ONLINE  * MySQL      * ONLINE  *
* SSH       * ONLINE  * Apache    * ONLINE  * Iptables   * ONLINE  *
* Fail2ban  * ONLINE  * IP Connect* ONLINE  * Ip6tables  * ONLINE  *
* BlueTooth * ONLINE  * Hidd      * ONLINE  * NTPD       * ONLINE  *
* Sendmail  * ONLINE  * Samba     * ONLINE  * Webmin     * OFFLINE *
* Ethernet0 * ONLINE  * Ethernet1 *   N/A   * Wlan0      *   N/A   *
********************************************************************
* Running Asterisk Version : Asterisk 1.4.21.2
* Asterisk Source Version  : 1.4.21.2
* Zaptel Source Version    : 1.4.12.1
* Libpri Source Version    : 1.4.7
* Addons Source Version    : 1.4.7
********************************************************************
pbx.local on 192.168.0.104 - eth0
CentOS release 5.2 (Final) :32 Bit Kernel: 2.6.18-92.1.6.el5

And the relevant part of the log is

-- Called g0/wTELNUMBER
    -- Zap/1-1 answered SIP/207-09817e08
    -- Hungup 'Zap/1-1'
  == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/207-09817e08' in macro 'dialout-trunk'
  == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/207-09817e08'
    -- Executing [h@macro-dialout-trunk:1] Macro("SIP/207-09817e08", "hangupcall|") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/207-09817e08", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/207-09817e08", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/207-09817e08", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/207-09817e08", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/207-09817e08' in macro 'hangupcall'
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/207-09817e08'

 

[blanchae]

The log is showing the zaptel group g0 answered a SIP call. The exact opposite of what you stated the problem is?

When dialing out through a ZAP channel (Digium X400), the call hangs up immediately when the recipient gets the phone.

 

[daniel..]

Thank you Eugene. I expressed myself wrong. Dialed out calls ring normally, but as soon as the call gets answered, the line shuts down.

 

[Astrosmurfer]

Check the channel's context in

/etc/asterisk/chan_dahdi.conf

context=from-zaptel

 

[daniel..]

Check the channel's context in

/etc/asterisk/chan_dahdi.conf

context=from-zaptel

Thank you. I do not have this file. I stayed with Asterisk 1.4 to avoid using dahdi (which I understand does not support the Digium TDM400). But FYI:

asterisk -rx "zap show channels"
   Chan Extension  Context         Language   MOH Interpret
 pseudo            default         es         default
      1            from-pstn       es         default
      2            from-pstn       es         default
      3            from-internal   es         default

Should it say "from-zaptel?"

 

[blanchae]

In /etc/asterisk/zapata-auto.conf and change the context to from-zaptel. Then from the Asterisk CLI>, type zaptel restart or whatever the command is to restart it. (help zaptel should give the answer)

See if that works.

 

[daniel..]

In /etc/asterisk/zapata-auto.conf and change the context to from-zaptel. Then from the Asterisk CLI>, type zaptel restart or whatever the command is to restart it. (help zaptel should give the answer)

See if that works.

Thank you. Tried that (changed /etc/asterisk/zapata-channels.conf) and it did not work.

Any other ideas will be greatly appreciated.

 

[blanchae]

Plug a POTS phone in the line and check that it is working properly on a normal call.