search for: nothingok

...E[3056] logger.c: -- Stopped music on hold on DAHDI/9-1 [Sep 29 07:18:17] DEBUG[3056] chan_sip.c: SIP transfer: Succeeded to masquerade channels. [Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: New owner for channel 8 is DAHDI/8-1 [Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: master: 8, slave: 9, nothingok: 0 [Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: Stopping tones on 8/0 talking to 9/0 [Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: Stopping tones on 9/0 talking to 8/0 [Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: Making 9 slave to master 8 at 0 [Sep 29 07:18:17] DEBUG[3218] chan_dahdi.c: Added 20 to...

Hello, I'm very new with * and I would really appreciate some help to implement a SIP to PSTN Gateway. My current scenario includes an * box with a TE405P board. I have a 1.5Mb connection to the outside world (using a router with firewall capabilities) and channel banks that allow me to connect the T1s coming out of the TE405 board to the PSTN network (carrier). I need to configure * to

...29 11:44:45] DEBUG[12424]: chan_dahdi.c:1797 dahdi_train_ec: No echo training requested [Oct 29 11:44:45] DEBUG[12424]: chan_dahdi.c:4440 dahdi_handle_event: channel 65 answered -- Zap/65-1 answered Zap/62-1 [Oct 29 11:44:45] DEBUG[12424]: chan_dahdi.c:3662 dahdi_bridge: master: 62, slave: 65, nothingok: 0 [Oct 29 11:44:45] DEBUG[12424]: chan_dahdi.c:3677 dahdi_bridge: Stopping tones on 62/0 talking to 65/0 [Oct 29 11:44:45] DEBUG[12424]: chan_dahdi.c:3689 dahdi_bridge: Stopping tones on 65/0 talking to 62/0 [Oct 29 11:44:45] DEBUG[12424]: chan_dahdi.c:3497 dahdi_link: Making 65 slave to master 62...

My office is investigating using an Asterisk PBX and also going to a VOIP provider for our main phone connections, but one of the tricky things is that we need to have outbound and inbound modem calls (fax too). I see a lot of talk about faxes but no mention of modems on this list. I get the impression that modems will be a problem for us. Is that true?

...21:08:42 DEBUG[2134]: chan_zap.c:1408 zt_enable_ec: No echo cancellation requested Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:1424 zt_train_ec: No echo training requested -- Attempting native bridge of Zap/3-1 and Zap/1-1 Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:3111 zt_bridge: master: 3, slave: 1, nothingok: 0 Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:3126 zt_bridge: Stopping tones on 3/0 talking to 1/0 Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:3138 zt_bridge: Stopping tones on 1/0 talking to 3/0 Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:2954 zt_link: Making 1 slave to master 3 at 0 Jun 6 21:08:42 DEBUG[21...

...1-8621<ZOMBIE> Feb 10 10:17:01 DEBUG[14917] channel.c: Bridge stops bridging channels SIP/570601-3ac4 and SIP/570601-8621<ZOMBIE> Feb 10 10:17:01 VERBOSE[14917] logger.c: -- Attempting native bridge of Zap/2-1 and Zap/1-1 Feb 10 10:17:01 DEBUG[14917] chan_zap.c: master: 2, slave: 1, nothingok: 0 Feb 10 10:17:01 DEBUG[14917] chan_zap.c: Stopping tones on 2/0 talking to 1/0 Feb 10 10:17:01 DEBUG[14917] app_dial.c: Exiting with DIALSTATUS=ANSWER. Feb 10 10:17:01 DEBUG[14917] chan_zap.c: Stopping tones on 1/0 talking to 2/0 Feb 10 10:17:01 VERBOSE[14917] logger.c: == Spawn extension (macr...

Hi list, I'm getting an error on a E1 link to the telco, after some weeks of operation without trouble. I have an asterisk with a TE405 in passtrough mode: two E1 are connected to the Telco, two E1 are connected to 2 Siemens PaBX. Only 15 channels are used on each E1 (conf is attached).The system has been in production for nearly a year, and does work flawlessly for weeks, then I

...19 ast_set_write_format: Set channel Zap/1-1 to write format ulaw Aug 4 15:44:13 DEBUG[5059]: channel.c:1752 ast_set_read_format: Set channel Zap/2-1 to read format ulaw -- Attempting native bridge of Zap/1-1 and Zap/2-1 Aug 4 15:44:13 DEBUG[5059]: chan_zap.c:2677 zt_bridge: master: 1, slave: 2, nothingok: 0 Aug 4 15:44:13 DEBUG[5059]: chan_zap.c:2692 zt_bridge: Stoping tones on 1/0 talking to 2/0 Aug 4 15:44:13 DEBUG[5059]: chan_zap.c:2704 zt_bridge: Stoping tones on 2/0 talking to 1/0 Aug 4 15:44:13 DEBUG[5059]: chan_zap.c:1263 zt_disable_ec: disabled echo cancellation on channel 1 Aug 4 15:44:...

Hello all I just checked out the latest zaptel/zapata/libpri/asterisk/asterisk-addons from the cvs and ran through the entire make procedures. Everything seemed to go fine however now when I attempt to start asterisk, it says ok but it seems to be immediately crashing. The following messages are displayed in my /var/log/asterisk/messages file for the time right around the crash: Dec 29

I currently have: Telco-PRI ---- Panasonic DBS576 PBX ---- E&M wink T1 ---- Asterisk. I have configured the Panasonic to forward my Asterisk DIDs to the Asterisk extensions over the T1. I do not get DID nor CID on the Asterisk, so I want to use PRI between the PBXs. I do not want to pay for another PRI card for the Panasonic. (T1 and PRI are different cards) I see this as my least