January 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] [Freeswitch-dev] Issue with Call Hold @ Freeswitch FXS....
by Irshad Ahamed 08 Jan '08

08 Jan '08

Your interpretation is right. This is what I have observed while testing here. To be more precise, when you hangup the FXS completely, the first call doesnt get disconnected. BR, Irshad. On Jan 8, 2008 7:38 PM, Anthony Minessale <anthmct(a)yahoo.com> wrote: > Can you clarify more so I am sure I understand? > > You use fxs to call an extension that leads to a sip phone. > You press FLASH on fxs to get a dialtone and make another call. > You hangup the fxs completely (ONHOOK) and one of the 2 calls do not end? > > The current intended behavior is when you FLASH fxs you can make > another call and switch back and forth with subsequent FLASH. But if > the fxs is ONHOOK all the calls should end. > > We still need some more code to do other behaviors from fxs like > transfer etc. > > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > > ----- Original Message ---- > From: "freeswitch-dev(a)lists.freeswitch.org" < > freeswitch-dev(a)lists.freeswitch.org> > To: freeswitch-dev(a)lists.freeswitch.org; > freeswitch-users(a)lists.freeswitch.org > Sent: Monday, January 7, 2008 11:36:09 PM > Subject: [Freeswitch-dev] Issue with Call Hold @ Freeswitch FXS.... > > Set-up is as below: > 1. Freeswitch is configured as UA with Asterisk TDM card (Openzap) 1 FXS + > 1 FXO registered with SIPX. > 2. SNOM300 registered with SIPX > > SNOM300: > Kernel Version:snom300 linux 3.25 > Application-Version:snom300-SIP 6.5.2 > Rootfs-Version:snom300 jffs2 v3.36 > Firmware-URL: > Production > Information:Mac:00041325329C;Version:Standard;Hardware:snom300 (MB > V3.2_A11);Date:26/06/06;Copyright(C) snom technology AG > 3. GRANDSTREAM registered with SIPX. > > GRANDSTREAM: > Product Model: GXP2000 > Software : Program-- 1.1.4.18 > Version : Bootloader-- 1.1.4.6 > > > Procedure followed: > 1. Make a successful call from Freeswitch FXS to SNOM300. Put the call on > hold @ Freeswitch FXS and make new call to GRANDSTREAM. > 2. Accept the call @ GRANDSTREAM and disconnect the connected call @ > FREESWITCH FXS. > 3. The second call with GRANDSTREAM is being disconnected but the first > call with SNOM300 is still alive at SNOM300. > > NOTE: Please find the Freeswitch logs and Ethereal caps @ > http://jira.freeswitch.org/browse/MODENDP-74 > > BR, > Irshad. > > > ------------------------------ > Looking for last minute shopping deals? Find them fast with Yahoo! Search.<http://us.rd.yahoo.com/evt=51734/*http://tools.search.yahoo.com/newsearch/c…> > > _______________________________________________ > Freeswitch-dev mailing list > Freeswitch-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > >

1 0

Freeswitch failover and distributed Voicemail + some debug propositions
by freeswitch-users＠lists.freeswitch.org 08 Jan '08

08 Jan '08

Hello, I have few questions. 1. Did anyone deployed freeswitch in failover mode via heartbeat? If yes what was the experiences, and some info? 2. Did anyone deployed freeswitch like a two-way distributed system, freeswitch on one server and freeswitch with voicemail on other? That way the main server would not need to work with voicemail, thus being a lot less under the load. 3. Anyone tested how much voicemail sessions can and on what machine be run simultaneously. Thanks, J.

1 0

Event Socket Questions
by Bill Binko 08 Jan '08

08 Jan '08

Ok, I have gotten the DTMF to work, and am much happier now. I have a nice framework forming that will make it easy for us to develop PHP-based IVR apps, monitors, etc. Live is looking better. However, I have some questions about the Event Socket interface that I would love some help on. First, it seems that when in 'async full' mode, you get ALL the events for any connections, not just the one you were invoked on (in outbound mode). If that's true, how do I discern which events are for my call? I suppose I could use the UUID in the Event, but that seems fairly wasteful - shouldn't I be able to just register for "my" events (that's why I'm calling "myevents" ;-) Second, what is the difference between the three ways to invoke functionality? There's the "basic" commands like "myevents", "event plain ALL", "log DEBUG". Then there are the api commands that are invoked like "api uuid_broadcast <UUID> <path>", and then there are the sendMsg command like "sendmsg\ncall-command: execute\nexecute-app-name: answer\nexecute-app-arg: \n\n" Each of these seem to invoke functionality in the current call (or the call with the given UUID), and there seems to be overlap (i.e. there is an 'answer' command in the API), but they all have different syntaxes, Content-Types and corresponding result messages. This isn't a complaint, I'm just trying to sort out when to use each. I'll be happy to update the wiki with the clarification. Finally, assuming this becomes reasonably usable, how do I add it to the contributed scripts? Thanks, Bill

2 1

Issue with Call Hold @ Freeswitch FXS....
by freeswitch-users＠lists.freeswitch.org 08 Jan '08

08 Jan '08

Set-up is as below: 1. Freeswitch is configured as UA with Asterisk TDM card (Openzap) 1 FXS + 1 FXO registered with SIPX. 2. SNOM300 registered with SIPX SNOM300: Kernel Version:snom300 linux 3.25 Application-Version:snom300-SIP 6.5.2 Rootfs-Version:snom300 jffs2 v3.36 Firmware-URL: Production Information:Mac:00041325329C;Version:Standard;Hardware:snom300 (MB V3.2_A11);Date:26/06/06;Copyright(C) snom technology AG 3. GRANDSTREAM registered with SIPX. GRANDSTREAM: Product Model: GXP2000 Software : Program-- 1.1.4.18 Version : Bootloader-- 1.1.4.6 Procedure followed: 1. Make a successful call from Freeswitch FXS to SNOM300. Put the call on hold @ Freeswitch FXS and make new call to GRANDSTREAM. 2. Accept the call @ GRANDSTREAM and disconnect the connected call @ FREESWITCH FXS. 3. The second call with GRANDSTREAM is being disconnected but the first call with SNOM300 is still alive at SNOM300. NOTE: Please find the Freeswitch logs and Ethereal caps @ http://jira.freeswitch.org/browse/MODENDP-74 BR, Irshad.

1 0

Re: [Freeswitch-users] SIP users
by Anthony Minessale 07 Jan '08

07 Jan '08

That approach is pretty good since it gives you a way to grant a certain extension to a certain user. We could also add an option to the sofia profile to insist that all users must use the same vaule for the username and the auth username. like <param name="inbound-reg-force-matching-username"/> This is less flexible but easier to setup since it does not require modification of every user in the directory. one or both of those solutions seem ok to me, let me know. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: David Knell <dave(a)3c.co.uk> To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, December 26, 2007 8:02:32 AM Subject: [Freeswitch-users] SIP users Hi all - Got a bit of an issue with registering endpoints - these being phones - with FS, which is that the username used for authentication is not necessarily the same as the username used for call routing. This is fine if you can trust your users (and their endpoints) to set them to be the same, but I can't. To be specific, a dialplan entry such as a bridge to sofia/sip.foo.com/2000%sip.foo.com will call whoever has set their SIP username to be 2000, which might be different to who's authenticated using an authentication username of 2000. Less wordily, any user can get any other user's calls by changing their SIP username to match that user's. I've added a few lines to src/mod/endpoints/mod_sofia/sofia_reg.c (see attached) to allow the username for an endpoint to be forced to be something, in the same way as sip-force-contact allows the contact to be set. A directory entry might now look like: <section name="directory"> <domain name="testing"> <user id="2000"> <params> <param name="password" value="password" /> </params> <variables> <variable name="sip-force-user" value="2000" /> </variables> </user> </domain> </section> A couple of questions. Firstly, have I overlooked something blindingly obvious here and am I barking up completely the wrong tree? Assuming not, is this the right approach, or should we - instead of forcing the username to be something - verify that it is that something and refuse the registration if not? Cheers -- Dave -----Inline Attachment Follows----- *** 344,349 **** --- 344,350 ---- int network_port; int cd = 0; const char *call_id = NULL; + char *force_user; /* all callers must confirm that sip, sip->sip_request and sip->sip_contact are not NULL */ switch_assert(sip != NULL && sip->sip_contact != NULL && sip->sip_request != NULL); *************** *** 419,424 **** --- 420,433 ---- char *exp_var; register_gateway = switch_event_get_header(*v_event, "sip-register-gateway"); + + /* Allow us to force the SIP user to be something specific - needed if + * we - for example - want to be able to ensure that the username a UA can + * be contacted at is the same one that they used for authentication. + */ + if ((force_user = switch_event_get_header(*v_event, "sip-force-user"))) { + to_user = force_user; + } if ((v_contact_str = switch_event_get_header(*v_event, "sip-force-contact"))) { if (!strcasecmp(v_contact_str, "nat-connectile-dysfunction") || !strcasecmp(v_contact_str, "NDLB-connectile-dysfunction")) { ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

4 4

Crash while incoming Call with improper dialplan/default.xml
by Irshad Ahamed 07 Jan '08

07 Jan '08

Hi, Crash has been found with Freeswitch when we configure the dialplan/default.xml as follows. <extension name="incoming"> <condition field="destination_number" expression="^(7001)$"> <action application="bridge" data="openzap"/> </condition> </extension> Procedures followed: 1. Configure for incoming call as above in your dialplan/default.xml 2. Run Freeswitch and register FXS with sipx. 3. After successful registration, make an inward call from other registered endpoint to Freeswitch FXS. It will result in a crash. BR, Irshad. On Jan 6, 2008 2:05 AM, Brian West <brian.west(a)mac.com> wrote: > Attention FreeSWITCHers, > As we get closer to b3 I would like to ask that everyone assist me > in > testing each and every application. As an open source project we have > very limited resources in testing so I call on the community to assist > me in testing. Please report any bugs to me, or the list.. and if you > feel like it post info to http://jira.freeswitch.org. It would be > greatly appreciated. > > This includes testing with valid, invalid and really invalid input > such as long lines.. garbage and such. > > I have started a wiki page for this. We should discuss automated > application testing scenarios if possible. > > http://wiki.freeswitch.org/wiki/Testing (bare page for now as I lab > up more tests and start logging them.) > > Thanks, > Brian West > FreeSWITCH > > _______________________________________________ > Freeswitch-dev mailing list > Freeswitch-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org >

1 0

DTMF Events not available on Event Socket (Outbound)?
by Bill Binko 05 Jan '08

05 Jan '08

I've been trying to implement a basic IVR that interacts with our database using PHP. With some help on IRC, I've created an Event Socket interface for PHP. It is rudimentary at this point, and not ready for prime-time, but it is working -- FreeSwitch connects to my process when a call comes in, sends the basic context info, and I can issue calls. After a call to "event plain ALL", I start receiving events. Mainly HEARTBEAT events, with a few others thrown in. What I'm NOT seeing is any DTMF events. Now, I know that the DTMF codes are recognized by other parts of FreeSwitch, such as the IVR, and I've ensure that my SIP client is sending them in-audio, SIP INFO and RFC2833. However, I'm not getting anything when I connect and press buttons. I have some other issues with the Event Socket, that I'd love to get through, but without the DTMF, most of the applications we're working on won't go very far. Is there anything I have to do to be able to get these Events to show up? Thanks Bill PS: I've attached the EventSocketListener PHP class. You can run it from the command line and it will listen on port 9090 -- obviously, it's not clean nor done, but it is working #!/usr/local/bin/php <?php /** * Based loosely on the NET_Server code in PEAR. * This is only an example - considerable additional work is needed * Specifically, the code in the handleConnection method should be * handled in a subclass * * */ class Message { var $properties = array(); var $content = null; } class EventSocketListener { var $host; var $port; var $sock; var $is_parent = true; var $clientInfo; var $clientFD; var $connectionContext = array(); function &create($port, $host = "localhost") { $esl = new EventSocketListener; $esl->port = $port; $esl->host = $host; if (!function_exists('socket_create')) { return PEAR::raiseError('Sockets extension not available.'); } return $esl; } function start() { if (($this->sock = socket_create(AF_INET, SOCK_STREAM, SOL_TCP)) === false) { echo "socket_create() failed: reason: " . socket_strerror(socket_last_error()) . "\n"; } if (!socket_set_option($this->sock, SOL_SOCKET, SO_REUSEADDR, 1)) { echo 'Unable to set option on socket: '. socket_strerror(socket_last_error()) . PHP_EOL; } if (socket_bind($this->sock, $this->host, $this->port) === false) { echo "socket_bind() failed: reason: " . socket_strerror(socket_last_error($this->sock)) . "\n"; } if (socket_listen($this->sock, 5) === false) { echo "socket_listen() failed: reason: " . socket_strerror(socket_last_error($this->sock)) . "\n"; } // Dear children, please do not become zombies pcntl_signal(SIGCHLD, SIG_IGN); // wait for incmoning connections while (true) { // new connection if(($fd = socket_accept($this->sock))) { $pid = pcntl_fork(); if($pid == -1) { return PEAR::raiseError('Could not fork child process.'); } // This is the child => handle the request elseif($pid == 0) { // this is not the parent $this->_isParent = false; // store the new file descriptor $this->clientFD = $fd; $peer_host = ""; $peer_port = ""; socket_getpeername($this->clientFD, $peer_host, $peer_port); $this->clientInfo = array( "host" => $peer_host, "port" => $peer_port, "connectOn" => time() ); $this->handleConnection(); socket_shutdown($this->clientFD, 2); socket_close($this->clientFD); } else /* Parent does nothing */ { } } } } function handleConnection() { $fd = $this->clientFD; //first, read headers & setup a state for this connection $line = ""; socket_write($fd, "CONNECT\n\n"); do { $line = socket_read($fd, 2048, PHP_NORMAL_READ); if (trim($line) == "") break; //we got a header, we need to add it to the list($key, $value) = explode(":", $line); $key = trim($key); $value = trim(urldecode($value)); $this->connectionContext[$key] = $value; } while ($line != ""); // print_r($this->connectionContext); $this->callConnected(); exit(); } function processMessages($returnOnReply = false) { $fd = $this->clientFD; $result = new Message(); $props = array(); while (true) { do { $line = @socket_read($fd, 2048, PHP_NORMAL_READ); if (socket_last_error($fd) == 104) return null; if ($line == null || $line == FALSE || trim($line) == "") break; //we got a header, we need to add it to the message list($key, $value) = explode(":", $line); $key = trim($key); $value = trim(urldecode($value)); $props[$key] = $value; } while ($line != ""); $result->properties = $props; if (isset($props['Content-Length'])) { $length = $props['Content-Length']; print("Reading content - $length\n"); $data = socket_read($fd, $length); $result->content = $data; } if (isset($props['Content-Type'])) { $type = $props['Content-Type']; if ($returnOnReply && ($type == "command/reply" || $type == "api/response")) { return $result; } else if ($type == "text/event-plain") //only plain events for now { $this->handleEvent($result); } } else { print("UNKNOWN MESSAGE: \n"); print_r($result); } } } function invokeCommand($command) { //Send the command print("Invoking: $command\n"); $this->sendCommand($command); // Wait for the response $result = $this->processMessages(true); return $result; } function sendCommand($command) { $fd = $this->clientFD; socket_write($fd, trim($command) . "\n\n"); } /*-----------------------------------------------------*/ /* Abstract Methods - should move to subclass*/ function callConnected() { print_r($this->connectionContext); print("----------------\n"); $result = $this->invokeCommand("log DEBUG"); print_r($result); $result = $this->invokeCommand("event plain ALL"); print_r($result); $this->processMessages(false); print("DONE PROCESSING MESSAGES"); print_r($this->connectionContext); } function handleCommandResponse($response) { print("Recieved Unhandled Response:\n"); print_r($response); } function handleEvent($event) { print("Recieved Unhandled Event:\n"); print_r($event); } } // create a server that forks new processes $server = &EventSocketListener::create(9090); // start the server $server->start(); ?>

3 5

Re: [Freeswitch-users] Not able to make an outgoing Call through FXO....
by Anthony Minessale 04 Jan '08

04 Jan '08

What does openzap.conf.xml look like? Perhaps you are using the default which only has 1 span defined. Try cloning the whole <span id="1"> block so there is a span 1 and 2 like this: <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> </settings> <analog_spans> <span id="1"> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/>     </span> <span id="2"> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/>     </span> </analog_spans> </configuration> Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Irshad Ahamed <irshad.fswitch(a)gmail.com> To: freeswitch-dev(a)lists.freeswitch.org; freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, January 2, 2008 12:40:14 AM Subject: [Freeswitch-users] Not able to make an outgoing Call through FXO.... Hi, Wishing U all a very Happy n Prosperous New Year!!!! I am trying to make an outgoing call using FXO connected to an external telephone line. Here is the set-up: 1. FXO connected to external telephone line 2. FXS connected to analog phone. The Openzap.conf @ /etc/openzap is as follows: [span zt] name => OpenZAP number => 1 fxs-channel => 1 [span zt] name => OpenZAP number => 1 fxo-channel => 4 The default.xml @ /usr/local/freeswitch/conf/dialplan is configured for inward and outgoing calls on FXO as follows: <extension name="outgoing-fxo"> <condition field="destination_number" expression="^(43)$"> <action application="bridge" data="openzap/2/1/43"/> </condition> </extension> <extension name="incoming-fxo"> <condition field="destination_number" expression="^(35)$"> <action application="bridge" data="openzap/1/1"/> </condition> </extension> While Start-up Freeswitch following logs were seen: 2008-01-02 11:51:22 [DEBUG] zap_io.c:1770 load_config() found config for span 2008-01-02 11:51:22 [DEBUG] zap_io.c:1797 load_config() created span 1 of type zt 2008-01-02 11:51:22 [DEBUG] zap_io.c:1810 load_config() span 1 [name]=[OpenZAP] 2008-01-02 11:51:22 [DEBUG] zap_io.c:1810 load_config() span 1 [number]=[1] 2008-01-02 11:51:22 [DEBUG] zap_io.c:1810 load_config() span 1 [fxs-channel]=[1] 2008-01-02 11:51:22 [DEBUG] zap_io.c:1850 load_config() setting trunk type to 'FXS' start(KEWL) 2008-01-02 11:51:22 [INFO] zap_zt.c:168 zt_open_range() configuring device /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:31 2008-01-02 11:51:22 [DEBUG] zap_io.c:1770 load_config() found config for span 2008-01-02 11:51:22 [DEBUG] zap_io.c:1797 load_config() created span 2 of type zt 2008-01-02 11:51:22 [DEBUG] zap_io.c:1810 load_config() span 2 [name]=[OpenZAP] 2008-01-02 11:51:22 [DEBUG] zap_io.c:1810 load_config() span 2 [number]=[1] 2008-01-02 11:51:22 [DEBUG] zap_io.c:1810 load_config() span 2 [fxo-channel]=[4] 2008-01-02 11:51:22 [DEBUG] zap_io.c:1839 load_config() setting trunk type to 'FXO' start(KEWL) 2008-01-02 11:51:22 [INFO] zap_zt.c:168 zt_open_range() configuring device /dev/zap/channel channel 4 as OpenZAP device 2:1 fd:32 2008-01-02 11:51:22 [INFO] zap_io.c:1881 load_config() Configured 2 channel(s) 2008-01-02 11:51:22 [CONSOLE] switch_loadable_module.c:748 switch_loadable_module_load_file() Successfully Loaded [mod_openzap] and while I made a call following logs were seen: 2008-01-02 12:04:16 [NOTICE] switch_core_state_machine.c:134 switch_core_standard_on_execute() Execute bridge(openzap/2/1/43)2008-01-02 12:04:16 [DEBUG] mod_openzap.c:231 tech_init() Set codec PCMU 20ms 2008-01-02 12:04:16 [NOTICE] switch_channel.c:392 switch_channel_set_name() New Chan OPENZAP/43 [e3630387-25d6-4031-bf81-239c3e64e3fa] 2008-01-02 12:04:16 [DEBUG] mod_openzap.c:802 channel_outgoing_channel() OPENZAP/43 State Change CS_NEW -> CS_INIT 2008-01-02 12:04:16 [DEBUG] switch_core_session.c:625 switch_core_session_signal_state_change() Kill OPENZAP/43 [BREAK] 2008-01-02 12:04:16 [ERR] zap_io.c:884 zap_channel_outgoing_call() outgoing_call method not implemented! 2008-01-02 12:04:16 [NOTICE] mod_openzap.c:818 channel_outgoing_channel() Close Channel OPENZAP/43 [CS_INIT] 2008-01-02 12:04:16 [ERR] switch_ivr_originate.c:525 switch_ivr_originate() Cannot Create Outgoing Channel! cause: DESTINATION_OUT_OF_ORDER 2008-01-02 12:04:16 [DEBUG] switch_ivr_originate.c:873 switch_ivr_originate() Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] 2008-01-02 12:04:16 [INFO] mod_dptools.c:1331 audio_bridge_function() Originate Failed. Cause: DESTINATION_OUT_OF_ORDER 2008-01-02 12:04:16 [NOTICE] mod_dptools.c:1358 audio_bridge_function() Hangup OpenZAP/43 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2008-01-02 12:04:16 [DEBUG] switch_channel.c:1071 switch_channel_perform_hangup() Kill OpenZAP/43 [KILL] 2008-01-02 12:04:16 [DEBUG] switch_core_session.c:625 switch_core_session_signal_state_change() Kill OpenZAP/43 [BREAK] 2008-01-02 12:04:16 [DEBUG] switch_core_state_machine.c:394 switch_core_session_run() (OpenZAP/43) State HANGUP 2008-01-02 12:04:16 [DEBUG] mod_openzap.c:329 channel_on_hangup() Changing state on 1:1 from IDLE to HANGUP 2008-01-02 12:04:16 [DEBUG] mod_openzap.c:359 channel_on_hangup() OpenZAP/43 CHANNEL HANGUP 2008-01-02 12:04:16 [DEBUG] switch_core_state_machine.c:45 switch_core_standard_on_hangup() Standard HANGUP OpenZAP/43, cause: DESTINATION_OUT_OF_ORDER 2008-01-02 12:04:16 [DEBUG] switch_core_session.c:683 switch_core_session_thread() Session 1 (OpenZAP/43) Locked, Waiting on external entities 2008-01-02 12:04:16 [INFO] switch_core_session.c:689 switch_core_session_thread() Session 1 (OpenZAP/43) Ended 2008-01-02 12:04:16 [NOTICE] switch_core_session.c:691 switch_core_session_thread() Close Channel OpenZAP/43 [CS_HANGUP] 2008-01-02 12:04:16 [DEBUG] zap_analog.c:351 zap_analog_channel_run() Executing state handler on 1:1 for HANGUP 2008-01-02 12:04:17 [DEBUG] zap_analog.c:290 zap_analog_channel_run() Changing state on 1:1 from HANGUP to BUSY 2008-01-02 12:04:17 [DEBUG] zap_analog.c:351 zap_analog_channel_run() Executing state handler on 1:1 for BUSY 2008-01-02 12:04:20 [DEBUG] zap_analog.c:664 process_event() EVENT [ONHOOK][1:1] STATE [BUSY] 2008-01-02 12:04:20 [DEBUG] zap_analog.c:690 process_event() Changing state on 1:1 from BUSY to DOWN 2008-01-02 12:04:20 [DEBUG] zap_analog.c:351 zap_analog_channel_run() Executing state handler on 1:1 for DOWN 2008-01-02 12:04:20 [DEBUG] mod_openzap.c:968 on_fxs_signal() got FXS sig [STOP] 2008-01-02 12:04:20 [DEBUG] zap_analog.c:648 zap_analog_channel_run() ANALOG CHANNEL thread ended. Could anyone help me out in getting FXO working ??? Thanks in advance... BR, Irshad. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

2 1

X100P
by Valentin Doroga 03 Jan '08

03 Jan '08

Has anybody tried using X100P on Windows? Should it work? Val.

3 2

irc chat
by Ivan C Myrvold 02 Jan '08

02 Jan '08

I have problems connecting to irc chat. I disconnected yesterday, and trying to connect today, but can't log in. What has happened? Ivan

7 17